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PortaBilling: User Manual

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1. The following instructions apply to Windows Messenger version 5 0 1 Start Windows Messenger and select Options from the Tools menu 2000 2006 PortaOne Inc All rights reserved www portaone com 127 Porta SIP Appendices File Actions Tools Help 4 My Add Contact Por Manage Contacts k E Click here Experience Sork Contacts By k Windows Messenger Oy x Manage Groups Cowork w Show Actions Pane W Custom Show Tabs Other Cr Use Windows Color Scheme Me kea All Cont lays on Top Audio Tuning Wizard Options I want to Add a Contact 2 Check the My contacts include users of a SIP Communication Service check box Enter your Sign in name as shown in the form ssername address where username is the name of the appropriate account in PB and address is either the IP address of the PortaSIP server or its name in DNS Then click the Advanced button Options Personal Phone Preferences Privacy Accounts Connection NET Passport Account 22 p ss Simedin net M Mu contacts include users of NET Messenger Service SIP Communications Service Account Sigmedin 8 M Mu contacts include users of a SIP Communications Service Sign in name 0118000 demo portaone cor Advanced Exchange 4ccount 2 2 AA NIP AA Not Signed In My contacts i
2. Configuration Profile Provision Enable yes Resync On Reset yes v Resync Random Delay 2 Resync Periodic 3600 Resync Error Retry Delay 3600 Forced Resync Delay 14400 Resync From SIP yes Resync After Upgrade Attempt yes Resyne Trigger 1 Resyne Trigger 2 Resync Fails On FNF yes M Profile Rule key B http PB_SLAVE_SERVER 4 M4 cfg Profile Rule B Profile Rule C Profile Rule D Log Resync Request Msg PN MAC Requesting resync SCHEME SERVIP PORT PATH Log Resync Success Msg PN MAC Successful resync SCHEME SERVIP PORT PATH Log Resync Failure Msg PN MAC Resync failed ERR Report Rule Firmware Upgrade Upgrade Enable Upgrade Error Retry Delay Downgrade Rey Limit Upgrade Rule Log Upgrade Request Msg PN MAC Requesting upgrade SCHEME SERVIP PORT PATH Log Upgrade Success Msg PN MAC Successful upgrade SCHEME SERVIP PORT PATH ER Log Upgrade Failure Msg PN MAC Upgrade failed ERR General Purpose Parameters GPP A GPP B 4 Enter the provisioning information e Inthe Profile Rule field enter key B http PB SLAVE SERVER S A SMA cfg replace PB_SLAVE_SERVER with the actual hostname or IP address of your PortaBilling100 slave server e Inthe GPP A field enter the numeric ID of the environment i_env e Ifyou plan to use encryption for the configuration files enter the secr
3. 3 Choose the type of connection PSTN to Vendor VoIP to Vendor etc by clicking on the corresponding tab 4 Press Add to add a new connection 5 Fill in the connection information If you send traffic to a vendor via telephony choose the node and enter the optional port pattern If you send traffic via VoIP enter the remote IP address address of the vendor s gateway or SIP server Choose the tariff which defines your termination costs for this connection vendor Description and 2000 2006 PortaOne Inc All rights reserved www portaone com 65 Porta SIP 8 Setting up SIP Services Capacity are mandatory for all connection types For VoIP connections where you have been assigned a login name and password choose the corresponding vendor account Click l Save Repeat steps 3 5 to add more connections to the same vendor then click Close to exit to the Vendor Management screen Repeat steps 2 7 to add connections for other vendors Create a Customer A customer is an account owner The customet s contact information is used to distribute generated accounts data and account usage information Even if your company owns and distributes all of its pre paid cards you will need at least one customer object for your company In the Management section of the Admin Index page choose Customers On the Customer Management page choose Add Fill in the New Customer form Please note that there are severa
4. 2000 2006 PortaOne Inc All rights reserved www portaone com 57 Porta SIP Setting up SIP Services E M SIP Phone subscribers tariff rates a t Boo Effective From Destination ou E Prati aroue Cour Country Interval sec Price USD min Effective From Edit Destination E 3 vrmmon ld fig delete Description First Next First Hext HH24 MI SS 1 1 0 23 0 23 immediately 1420601 a i Off Peak 1 1 0 20 ozo Sono 0 12000 0 12000 2005 08 02 E CZECH REPUBLIC Peak 0 11000 0 11000 11 50 55 420 Proper Off Peak E CZECH REPUBLIC Peak 4202 0 11000 0 11000 2005 08 02 Prague Off Peak 0 06000 0 06000 11 51 27 1 1 1 1 1 1 1 1 1 On the Tariff Management page you will see a list of the available tariffs Click the Rates icon next to the name of the tariff When you are in Tariff Management for a particular tariff click on Rates in the toolbar 2 On the Edit Rates screen click 4 Add 3 Fill in the required information o Destination destination prefix may be entered directly e g 47 for Norway or you can access the destinations directory by clicking the Destination link in the column header Here you can find the desired prefix by country name NOTE The phone prefix you are trying to create a rate for must already exist in Destinations Interval First first billing unit in seconds Interval Next
5. f ToDate 2005 07 29 YYYY MM DD Show Incomplete Calls Show CDRs Call history gt cose a Download E cors obets Account 1206001236 Balance 9 88500 USD Total Charged Time min sec 1 18 Opening Balance 10 00000 USD Total Charged 0 11500 USD Charged by Example SIP services product Total Credits 0 USD Type Debit Total Transactions 1 From 2005 07 28 00 00 00 To 2005 07 30 00 00 00 View From To Country Y Description Date Time Charged time min sec Amount USD E 1206001236 16044680035 North America 2005 07 29 12 45 36 1 18 0 11500 Login with the account s web access login and password After that you will be able to see the account s self care menu Choose the date range for which you want to see a list of calls and press Show CDR In the results table you will see call charges and other fees such as maintenance fees or refunds if any The report can be also downloaded by clicking the amp Download csv icon Check the Call History If you want to see a list of all calls going through the system or perhaps only ones for a particular destination use the Trace Call function 2000 2006 PortaOne Inc All rights reserved www portaone com 9 9 Porta SIP Setting up SIP Services HR Call Trace Destination 1 10 min H323 conf id E From 2005 07 29 YYYY MM DD 00 00 HH Mi To 2005 07 30 YYYY MM D
6. 1 Open the Account Info form for an existing account and go to the Additional Info tab E M Account Info Retail Customer John Doe Account ID 120612 34560 i Product USD SIP Subscribers Blocked F Balance 0 00000 USD User Agent Contact Preferred Language en English v Redirect Number UM Enabled CO IP Phone Sipura IP Phone Port Follow Me Enabled E commerce Enabled Discount Plan Product s default Music On Hold Customer s default 2 Select the IP Phone from the list 3 Choose specific port from the IP Phone Port select menu if the device has multiple phone ports 4 Click el Save amp Close 2000 2006 PortaOne Inc All rights reserved www portaone com 99 Porta SIP Setting up SIP Services Note The IP Phone select field shows a list of phones that have not been used before in other accounts or phones with available unused ports After the automated script creates a new configuration file it will place it on the provisioning server The file generation process runs every hour or every time you update a specific IP phone profile All device configuration files are stored in home porta admin apache htdocs on the PortaBilling100 slave server in subdirectories with a name identical to the environment ID 1_env 1 2 3 and so on IP phone device configuration If your IP phone ATA was pre configured
7. Acct Session Id h323 remote address h323 session protocol sipvZ h323 setup time 06 37 59 000 GMT Fri Jul 22 2005 h323 voice quality 0 Acct Terminate Cause h323 disconnect time h323 connect time User Request 06 38 24 000 GMT Fri Jul 22 20057 D6 358 07 000 GMT Fri Jul 22 20057 Acct Session Time 11 7 h323 disconnect cause 0 Acct Status Type Stop RTP session is closed 21 Jul 23 38 24 GLOBAL rtpproxy received command D lefdc57b 6063c200192 168 1 180 5qd911c473909f6d59b54af2f14f12986 813da13c69239f8d427bbd3b78c601ee 2000 2006 PortaOne Inc All rights reserved www portaone com g 4 Porta SIP Setting up SIP Services 21 Jul 23 38 24 1lefdc57b 80c3c2c 192 168 1 180 rtpproxy deleting session 0 on ports 35134 35136 forcefully 21 Jul 23 38 24 1efdc57b 80c3c2c 192 168 1 180 rtpproxy RTP stats 574 in from caller in from callee 538 1112 relayed 0 dropped 21 Jul 23 38 24 1efdc57b 80c3c2c 192 168 1 180 rtpproxy RTCP stats 3 in from callee ports Incoming call leg is closed 21 Jul 23 38 24 GLOBAL ser O in from caller 21 Jul 23 38 24 1lefdc57b 80c3c2c 192 168 1 180 rtpproxy 35134 35136 is cleaned up 21 Jul 23 38 24 GLOBAL rtpproxy 3 relayed 0 dropped session on sending reply 0 SENDING message to 62 244 32 30 50563 BYE sip 18667478647758Q062 244 32 30 50563 SIP 2 0 Via SIP 2 0 UDP 70 68 0 213 branch z9hG4
8. Convert ANI CLI for incoming calls into this dialing format Check yourself To call 1234567 outside of your office but within the same area 3 1234567 you dial To call long distance 5 1234567 within your country you dial 151234567 To call 1405 1234567internationally you dial 011 1 405 1234567 So when one of the accounts of this customer tries to make a call to 90042021234567 the SIP server will send an authentication request to the billing Billing can apply this customer s translation rule if defined or node translation rule if defined 2000 2006 PortaOne Inc All rights reserved www portaone com 70 Porta SIP Setting up SIP Services Set up Abbreviated Dialing for the Customer optional If your customer has multiple SIP accounts and plans to make calls between them it would be very inconvenient to have to dial a complete E 164 number each time Therefore you can create abbreviated dialing rules so that it will suffice to dial for example 120 to reach a Jeff Smith from any SIP phone using the customer s account 1 In Abbreviated Number Length enter the maximum number of digits in the abbreviated number e g if you plan to have extension numbers 401 402 and so forth the length will be 3 Click lal Save 2 Nowa table of abbreviated numbers will appear Click on Add to add a new extension 3 Enter the abbreviated number and the actual phone number the call will be forwar
9. DestinatiorDestinatior Country Descriptior First Interv Next Intery First Price Next Price Offpeak F Off peak N Off peak F Off peak N Forbi Hidde Discontinu Effective Fror CZECH REProper 30 6 0 15 0 15 30 6 0 1 0 1 5 11 2004 3 420602 CZECH RE Mobile 30 6 0 17 0 17 30 6 0 15 0 15 5 11 2004 3 an iba File Edit View Insert Format Tools Data Window Help Edit the file by adding more rows with rate data so that it resembles the screenshot below Note that the Country and Description columns are only for reference and are ignored during import Also when using the default template you must fill in data in the Off peak columns even if your tariff does not have an off peak period use the clipboard to easily copy the values from the 4 peak columns Also note that you may only use those phone prefixes which you already have defined as destinations see the Create destinations step above Make sure that you clear the values in the Effective from column which would mean that the new rates are effective immediately or enter a future date there Otherwise if you retain past dates these rates will fail to upload Off peak P Destinatior Free Seco Post Call Login Fee Connect Fee startstop hr 20 5 0 0 0 Destinatior Destinatior Country Descriptior First Interv Next Interv First Price Next Price Off peak F Offpeak N Off peak F Offpeak N Forbi Hidde Discontinu Effective Fror CZECH REProper 30 6 0 15 0 15 30 420602 CZECH RE
10. Cheap 6 0 025 This vendor has the same preference as vendor C but a L per minute rate CO Cheap 6 0 03 Expensive 5 0 11 This is the only eee in the last route category Vendor F was not included in the routing since his route category is not in the customer s routing plan i 4 Test Dialplan gt cose 4 Objects Bogut Protocol 5 Date and Time H323 SIP WY MM DD HH Mi 8610234567 Standara 5 0 mM il i Phone Number Routing Plan Destination Country Description Price Route Category Preference Huntstop Route to Vendor Connection Tariff 1 86 CHINA Proper 0 06000 USD Default 5 N ny gw 01 vendor B Termination to carrier B VYendorB 2 8610 CHINA Beijing 0 04000 USD Cheap T N ny gw 01 vendor A Termination to carrier A vendor A 3 86 CHINA Proper 0 02500 USD Cheap 6 N 45 12 156 200 vendor D Termination to vendor D vendor D 4 86 CHINA Proper 0 03000 USD Cheap 6 N 5 86 CHINA Proper 0 11000 USD Expensive 5 N ny gw 01 vendor C Termination to carrier C Vendor C 193 50 123 6 YendorE Termination to carrier E vendor E Number translation There are many different phone number formats some used by your customers others by your vendors How to deal with all of them without making mistakes PortaBilling offers a powerful tool called translation rules for converting phone numbers with several different types depending on customers needs Your network numbering plan The key to avoiding problems with
11. e Add the appropriate rate to the tariff associated with the accounts to be charged For example if you have SIP accounts with the prefix 078 then you should add the appropriate rate for destination 078 to the tariff used to charge for outgoing calls e Create a special tariff with rates corresponding to the prefixes allocated for your SIP accounts 078 in the example above This will be the tariff used to calculate your termination expenses Since you do not pay anything for such termination you can enter zero prices for all of the rates e Create a new vendor with a descriptive name for example Direct termination to SIP phones Add a VoIP to Vendor connection to that vendor with the tariff created in the previous step and enter sip ua in the Remote IP field So now if a call is made from one SIP phone to another the originating party will be charged according to the rates you have entered in the customer s tariff This call will be counted as terminated to the vendor Direct termination to SIP phones with zero termination cost but it will still be recorded in the database so you can easily view statistics for all SIP SIP calls bill incoming calls from PSTN to SIP using a special rate In order to properly bill a SIP account for such calls do the following e Install a PSTN2SIP application on your Cisco gateway which handles incoming PSTN calls e Create an appropriate tariff with the desired rates For examp
12. remote ip None and CLD 380443333333 Jul 21 22 38 24 Unknown node IP or no VoIP from vendor connections for this node Jul 21 22 38 24 Connection to vendor not found Jul 21 22 38 24 Connection to vendor not found on net call leg This is an on net call leg while the call is still traveling on our network so it is ignored Jul 21 22 38 24 Jul 21 22 38 24 Accounting response DOR Verify Call History for the Account To view the CDR of an account go to Customers select the customer owning the account and click on the Accounts icon or alternatively 91 Porta SIP Setting up SIP Services select the Account Info link from the Main Menu You can also go to the account self care page accessible via the Accounts menu item in the Home popup menu E 4 Accounts of Retail Customer CallSmart Ltd fe E add E account Generator close Eaves B toot SIP Status v Show Accounts CDRs AccountiD Idle days Currency Balance Credit Limit Type Product Batch Status SIP 1206001234 6 USD 10 00000 Debit Example SIP services demo A 1206001235 36 USD 9 69300 Debit Example SIP services 1206001236 4 USD 9 88500 Debit Example SIP services 1206001238 USD 10 00000 100 00000 Credit Example SIP services 1206001237 6 USD 9 94300 Debit Example SIP services E 1206001239 USD 75 00000 Debit Mexample Call history Pr aose a owes dd o Erom Date 2005 07 28 YYYY MM DD
13. Radius shared secret and so Of i 4 Node Management aaa E ce mi toot A E H323 ID IP Radius Client Oves Ono all a Name h323 id IP Radius Client Delete a E o A Add Node gt ia sove ia Save n e cose tot a a E Node Name Demosip a E a Node info E p E h323 id demosip mydomain com Manufacturer PortaOne x E E h323 password cisco R Type PortasIP M m NAS IP Address 207 52 37 56 l E Hostname z Domain Radius Client M Auth Transl Rule Radius Key phenttin j a m Radius Source IP 207 52 37 56 f m Radius Dictionary cisco a Submitted information is being cached in the billing engine and will not take effect immediately Default caching time is 10 minutes Please contact your system administrator for more information 1 Inthe management section of the Admin Index page choose Nodes 2 In the Node management window click the l Add icon 2000 2006 PortaOne Inc All rights reserved www portaone com 5 4 Porta SIP Setting up SIP Services 3 Fill in the New Node form o Node name a short descriptive name for your SIP server this will be used in the select menus o H323 ID recommended hostname domainname o H323 Password if you plan to send calls from your SIP server to your Cisco gateways where the default Cisco remote IP authentication script will be used enter cisco here o
14. communicate with a single central billing which provides all the required service provisioning information and maintains a global database of SIP phone registrations A SIP phone user may connect to any of the available PortaSIP servers only those which are available to him via his product s accessibility of course Once a SIP phone is successfully registered to one of the SIP servers the information is globally available within this PortaSwitch environment Porta K Billing Billing Engine Billing SIP Provisioning Registrations PortaSIP By installing several independent PortaSIP servers you can achieve two main goals e Improve the reliability of your network e Optimize call flow on your network so as to better utilize the available network infrastructure 2000 2006 PortaOne Inc All rights reserved www portaone com 99 Porta SIP System Concepts Improved reliability Porta 4 Billing Billing SIP Provisioning Registrations PortaSIP PortaSIP PortaSIP Even if one of the SIP servers is down due to network issues or hardware problems your subscribers can continue using the service via other SIP servers Billing Engine Better network utilization You can install several SIP servers in different geographical locations as shown below with users within a certain network able to use the closest available SIP server So if user A from Singapore calls user B also from Singapore the c
15. communication services at unusually low initial and operating costs that cannot be matched by yesterday s circuit switched and narrowband service provider PSTN networks In addition to conventional IP telephony services PortaSIP provides a solution to the NAT traversal problem and enhances ITSP network management capabilities It can be used to provide residential business and wholesale traffic exchange services 2000 2006 PortaOne Inc All rights reserved www portaone com g Porta SIP System Concepts PortaSIP functions a Termination Termination mn ay partner partnerB Bank Online payment processor Porta Billing y Administrator gt eR L interface Porta 4 SIP aa SS ee Residential IP Admin Web m Self care lt SS ANI DNIS Termination Pre paid cards Callback to PSTN Customized IVR Porta amp UM ee Unified Messaging Phone amp Web Interface PortaSIP provides the following functionalities e SIP registration allowing SIP phones to use the service from any IP address static or dynamically assigned e Customizable greeting upon successful service activation e Authorization for all incoming calls e Customer numbering plans to ensure correct phone number translation e Facilitation of communication between SIP phones behind a NAT e Error announcements from the media server e Automatic disconnect of calls when the maximum credit time is reached e Au
16. or on a public IP address if the device is behind a NAT firewall the administrator will not be able to access it 2000 2006 PortaOne Inc All rights reserved www portaone com 47 Porta SIP System Concepts Due to these reasons and since every device must be provisioned individually this method is acceptable for a testing environment or small scale service deployment but totally inappropriate for ITSPs with thousands of IP phones around the world Auto provisioning This approach is a fundamentally different one Instead of attempting to contact an IP phone and change its parameters pop method the initiative is transferred to the IP phone itself The device will periodically go to the provisioning server and fetch its configuration file Cisco ATA Expert This utility allows you to simplify manual provisioning of a Cisco ATA 186 188 device and browse the device configuration in a user friendly format For example instead of entering values such as 0 1 2 3 etc for gt 2 gt codec selection you can choose names such as G 729 from a select menu This tool is convenient for single time configuration of a device or for troubleshooting IP Phone Provisioning When you use auto provisioning for an IP phone instead of entering the same values for codec server address and so on into each of a thousand user agents you can simply create a profile which describes all these parameters Then PortaBilli
17. ot 729 codec since in this case you must pay a license fee A solution is to pre convert this voice prompt into a 8723 or 9729 byte stream store it in a file with the same name but with the 723 or 729 extension and upload it to PortaSIP The media server will then use the appropriate file calculate how much bandwidth need for my PortaSIP server The amount of bandwidth required for SIP signaling is insignificant compared to that used by the RTP stream so the most important task is to correctly estimate your RTP bandwidth needs of course this is only applicable if an RTP proxy is used otherwise the voice stream goes directly between the SIP phone and the remote gateway The http www voip info org wiki Bandwidth consumption website provides information regarding bandwidth consumption by voice calls depending on the codec used Do not use the codec bit rate in your calculations but rather an actual bandwidth figure which takes IP headers into account For example if you anticipate a maximum of 60 simultaneous calls with the 720 codec you will need 31 2Kpbs 2 60 3 7Mbps Note that we 2000 2006 PortaOne Inc All rights reserved www portaone com 1 1 0 Porta SIP How to multiply the one call bandwidth not just by the total number of calls but also by 2 since every call will be coming both in and out of the RTP pr OXY enable my SIP phone or ATA to be automatically pro
18. unlock Jul 22 2005 Jul 22 2005 If we have not received acknowledgement of call termination from the caller it will be resent 21 Jul 23 38 25 GLOBAL ser SENDING message to 62 244 32 30 50563 BYE sip 18667478647758Q062 244 32 30 50563 SIP 2 0 Via SIP 2 0 UDP 70 68 0 213 branch z9hG4bK4e23 398ad42b5b3be3602617bd8b10be8c17 0 Via SIP 2 0 UDP 70 68 0 213 5061 branch z9hG4bK4de91149a7529218ce70fb9209be2ef9 rport 5 061 Max Forwards 16 2000 2006 PortaOne Inc All rights reserved www portaone com 89 Porta SIP Setting up SIP Services From lt sip 380443333333 70 68 0 213 gt tag 54cce4d15b86cbhbe9dec50522083bflc To 758 lt sip 1866 74 78647 758670 68 0 213 gt tag a4044d2fe886380800 Call ID 1l1efdc57b 80c3c2c 192 168 1 180 CSeq 100 BYE Contact Anonymous lt sip 70 68 0 213 5061 gt Expires 300 User Agent Sippy cisco GUID 2619723805 2664678018 2210055510 2363514890 h323 conf id 2619723805 2664678018 2210055510 2363514890 The caller s SIP UA confirms call disconnection 21 Jul 23 38 25 GLOBAL ser RECEIVED message from 62 244 32 30 50563 SIP 2 0 200 OK To 758 lt sip 18667478647758 7 70 68 0 213 gt tag a4044d2fe886380800 From lt sip 380443333333 70 68 0 213 gt tag 54cce4d15b86cbhbe9dec50522083bflc Call ID 1l1efdc57b 80c3c2c 192 168 1 180 CSeq 100 BYE Via SIP 2 0 7UDP 70 68 0 213 branch z9hG4bK4e23 398ad42b5b3be3602617bd8b10be8c17 0 Vias S1P 2 0 UDE 70 68 0 213 50
19. voice class codec 1 session protocol sipv2 dtmf relay rtp nte fax protocol cisco configure my Cisco ATA186 to work with PortaSIP Perform the initial network configuration of the ATA using the built in IVR After your ATA is assigned an IP address you can go to the web configuration screen at http lt your ATA I P address gt dev Consult APPENDIX C Clients Cisco ATA 186 Configuration for PortaSIP For other options not listed in the table below the default manufacturer value is assumed provide services to and bill a customer who has a SIP enabled gateway but no authorization capability e g Cisco AS5350 PortaSIP is able to authenticate incoming calls using the IP address of the remote side This method ensures that PortaSIP will accept calls from yout own gateways but it can also be used to bill traffic from your customers You just need to create an account for your customer with an account ID identical to the IP address of his gateway Authentication and billing will be done in the same way as IP based billing using H323 make all SIP calls to a certain prefix NNN go to my gateway XXX Normally it is only possible to use the REGISTER command for user agents i e for devices which represent a single physical phone An SIP user agent cannot register with the SIP server and report I am going to receive all calls for prefix NNN Cisco 5300 supports the REGISTER command but this only works for numb
20. 2006 PortaOne Inc All rights reserved www portaone com 43 Porta SIP System Concepts aware of its external mapped address and port and can insert them into the contact information as the IP port for receiving SIP messages You should be careful to use a registration interval shorter than the keep alive time for NAT mapping RTP Media Stream An RTP that must traverse a NAT cannot be managed as easily as SIP signaling In the case of RTP the SIP message body contains the information that the endpoints need in order to communicate directly with each other This information is contained in the SDP message The endpoint clients fill in this information according to what they know about themselves A client sitting behind a NAT knows only its internal IP port and this is what it enters in the SDP body of the outgoing SIP message When the destination endpoint wishes to begin sending packets to the originating endpoint it will use the recetved SDP information containing the internal IP port of the originating endpoint and so the packets will never arrive Understanding the SIP Server s Role in NAT Traversal Below is a simplified scheme of a typical SIP call SIP Server Jum Ee g Media RTP g UA 1 UA 2 It must be understood that SIP signaling messages between two endpoints always pass through a proxy server while media streams usually flow from one endpoint to another directly Since the SIP Server is
21. 29 4 disable v G 711 RTP payload type Dynamic Type 126 127 c 729a Ml Standard Type 0 8 Phone 2 G 711 Silence Supression G 711 Only oO DTMF Relay by negotiation Hookflash Relay disable x 4 Press l Save to save the new configuration to the ATA Testing the Whole System Make sure the PortaBilling radius and PortaSIP servers are running 2 Configure your SIP user agent with the account ID and password See appendices for configuration guidelines for some SIP UAs Then have your SIP user agent login to the SIP server 3 Check that the account is logged into the SIP server O Goto the account list screen and see if the SIP indicator button a blue circle is on for this account 2000 2006 PortaOne Inc All rights reserved www portaone com 73 Porta SIP Setting up SIP Services Hz Accounts of Retail Customer EasyCall Ltd Pe E ada E Account Generator coe tooo Account ID Batch Ctri SIP Status au zji a v Chorus _ CDRs AccountiD Idle days Currency Balance Credit Limit Type Product Batch Status SIP 12061234567 4 USD 10 00000 Debit SIP Subscribers easycall Fe 12061234568 4 USD 10 00000 Credit SIP Subscribers easycall 12061234569 4 USD 10 00000 Dehit SIP Subscribers easycall E 12061234570 4 USD 0 00000 Credit SIP Subscribers easycall O Go to the account info page for this account and check that
22. 3 The Access code or Info Digits fields only make sense when a call originates from your customer in a public telephony network Therefore just leave this empty for the SIP service 4 Click lel Save to save this accessibility entry 5 Repeat steps 1 4 if you want to define more accessibilities Make sure that you have a row in Accessibility containing the PortaSIP server and the tariff you want to use for outgoing SIP calls Create Vendors This step is only required if you have not entered information about your vendors into the system before Vendors are your termination partners or providers of incoming toll free lines 1 Inthe Management section of the admin interface choose Vendors 2 On the Vendor Management page choose 4 Add 2000 2006 PortaOne Inc All rights reserved www portaone com 63 Porta SIP Setting up SIP Services i 4 Add Vendor gt id save fal saverclose coef gt Vendor Name GlobalNet F Currency usp US Dollar v Opening Balance o Categorizing and Defaults Billing Period Daily x Bi lateral Traffic Exchange Offset Balance with Customer NONE v Minimum Amount to Offset E 4 Edit Vendor gt fel save i Save amp Close cos Connections Ut Log Vendor Name Globalnet E Opening Balance 0 00000 USD Balance 0 00000 USD Address Info Additional Info User Interface Accounts Notepad Login Time Zone Ameri
23. 62 244 32 30 From 758 lt sip 18667478647758 70 68 0 213 gt tag a4044d2fe886380800 To lt sip 380443333333 70 68 0 213 gt Call ID lefdc57b 80c3c2c 192 168 1 180 CSeq 101 INVITE Server Sip EXpress router 0 9 0 i386 freebsd Content Length 0 Request is sent to B2BUA 21 Jul 23 37 26 GLOBAL ser SENDING message to 70 68 0 213 5061 INVITE S107300443333333070 68 0 21325061 SIP 2 0 Record Route lt sip 7 0 68 0 213 ftag a4044d2fe886380800 1r gt Via SIP 2 0 UDP 70 68 0 213 branch z9hG4bK5e23 148da7017688248e9f2a16b450895a5b 0 Via SIP 2 0 UDE 192 168 1 180 5060 rport 50563 received 62 244 32 30 branch z 9hG4bK 6ad8150b From 758 lt sip 18667478647758 70 68 0 213 gt tag a4044d2fe886380800 To lt sip 380443333333 70 68 0 213 gt Call ID 1l1efdc57b 80c3c2c 192 168 1 180 CSeq 101 INVITE Max Forwards 16 Contact 758 lt sip 18667478647758 62 244 32 30 50563 gt Expires 240 User Agent Sipura SPA2000 2 0 13 g Content Length 467 Allow ACK BYE CANCEL INFO INVITE NOTIFY OPTIONS Supported x sipura Content Type application sdp PortaBilling Notify NAT v 0 o 2284 2284 IN IP4 192 168 1 180 a c IN IP4 62 244 32 30 t 0 0 m audio 16384 RTP AVP 18 0 2 4 8 96 97 98 100 101 a rtpmap 18 G729a 8000 a rtpmap 0 PCMU 8000 a rtpmap 2 G726 32 8000 a rtpmap 4 G723 8000 a rtpmap 8 PCMA 8000 a rtpmap 96 G726 40 8000 2000 2006 PortaOne Inc All rights reserved www por
24. Checklist Print the following page and use it to check off operations you have completed while performing system setup according to the instructions in this chapter Please make sure that you perform all of the operations all of the boxes must be checked otherwise the service will not work Operation Done General configuration Fill in company data in Company Info Specify base currency For any other currency you plan to use specify the exchange rate source and define exchange rates Create all required destinations for off net calls Create a destination for your SIP numbers Network configuration Create a node for your PortaSIP Rating configuration Create a tariff A which will be applied to SIP subscribers Enter rates in tariff A for the destinations you plan to call both off net and SIP to SIP calls Create tariff B which describes your termination costs and routing for off net calls make sure it has a Routing type Enter rates in tariff B for the destinations you plan to call Create a tariff C which describes your termination costs for SIP to SIP calls Enter rates in tariff C for SIP destinations zero cost Create your SIP product Create one accessibility entry for this product using the PortaSIP node and tariff A Create an off net calls vendor Create a connection for this vendor using tariff B Create a fake SIP to SIP vendor Create a VoI
25. Concepts gateway in Moscow Russia the rule will be strip leading 810 or replace leading 8 with 7 Since customer based translation rules were introduced node based translation has become obsolete Therefore a node based translation rule is applied only if there is no customer based translation rule defined for a given customer Number translation on your network Below is an illustration of how different translation rules are applied during a call Porta Billing100 customer rate amp routing connection dialing rule lookup outgoing rule 0042021234567 42021234567 01142021234567 ROUTING IP address 1 2 3 4 01142021234567 AUTHORIZATION 0042021234567 Porta SIP 0042021234567 01142021234567 Number inside of your VoIP is represented as 42021234567 Carrier ABC Customer s IP Phone Cell Phone 1 The customer dials a phone number on his SIP phone He enters the number in the same format he uses on his conventional phone 1 e 0042021234567 2 The number is delivered to the PortaSIP server and translated using the customer s dialing rule which states that the international dialing prefix for this customer is 00 So the number becomes 42021234567 E 164 format This number is used to seatch for the customet s rate for this destination 3 PortaSIP then requests routing for this number Carrier ABC is defined for terminating calls to the Czech Republic in
26. Expires 240 User Agent Sipura SPA2000 2 0 13 g Content Length 467 Allow ACK BYE CANCEL INFO INVITE NOTIFY OPTIONS REFER Supported x sipura Content Type application sdp PortaBilling Notify NAT v 0 o 2284 2284 IN IP4 192 168 1 180 e c IN IP4 62 244 32 30 t 0 0 m audio 16384 RTP AVP 18 O 2 4 8 96 97 98 100 101 a rtpmap 18 G729a 8000 a rtpmap 0 PCMU 8000 a rtpmap 2 G726 32 8000 a rtpmap 4 G723 8000 a rtpmap 8 PCMA 8000 a rtpmap 96 G726 40 8000 a rtpmap 97 G726 24 8000 a rtpmap 98 G726 16 8000 a rtpmap 100 NSE 8000 a rtpmap 101 telephone event 8000 a fmtp 101 0 15 a ptime 30 a sendrecv a direction active a oldmediaip 192 168 1 180 B2BUA sends authorization request to the billing 21 Jul 23 37 26 lefdc57b 80c3c2c 192 168 1 180 b2bua sending AAA request User Name 1866 7478647758 Digest Realm 70 68 0 213 Digest Nonce 5dbe8444a6063753F35c247b356641ca Digest Method INVITE 2000 2006 PortaOne Inc All rights reserved www portaone com 78 Porta SIP Setting up SIP Services Digest URI sip 380443333333070 68 0 213 Digest Algorithm MD5 Digest User Name 1866 7478647758 Digest Response 7 f3a6fffcblf68b560a20937343e3ef0 Calling Station Id 18667478647758 Called Station Id 380443333333 h323 conf id call id 9C25D01D 9ED3C282 83BAC556 8CEQ600A l1efdc57b 80c3c2c 192 168 1 180 h323 remote address None h323 session
27. Mera Systems This call leg crosses a connection to the vendor Jul 21 22 38 24 Found vendor connection Jul 21 22 38 24 Charging call Jul 21 22 38 24 Calculating account s charge by tariff Porta SIP Billing calculates how much the account should be charged for this call Jul 21 22 38 24 Using peak rate since no off peak is defined Jul 21 22 38 24 PrepareNexecute GetPricePerDestination Jul 21 22 38 24 SQL query GetPricePerDestination executed in 0 005341 seconds Jul 21 22 38 24 Call to 380443333333 with duration 17 seconds will be charged for 30 seconds and cost is 0 0803 1x30x0 1606 5 by rate 125476 Jul 21 22 38 24 Calculating vendor s charge by tariff Mera Systems Also billing calculates the termination costs for this call Jul 21 22 38 24 Using peak rate since no off peak is defined Jul 21 22 38 24 PrepareNexecute GetPricePerDestination Jul 21 22 38 24 SQL query GetPricePerDestination executed in 0 005334 seconds Jul 21 22 38 24 Call to 380443333333 with duration 17 seconds will be charged for 30 seconds and cost is 0 0475 1x30x0 095 by rate 125321 Jul 21 22 38 24 Charging account for the call Jul 21 22 38 24 Inserting CDR CDRs are entered and balances are modified Jul 21 22 38 24 PrepareNexecute InsertAccountCDR Jul 21 22 38 24 Charging credit account 18667478647758 0 0803 Jul 21 22 38 24 PrepareNexecute UpdateAcc
28. PortaSIP receives an incoming VoIP call and sends an authorization request to the billing 3 e The billing detects that this call is coming via a VoIP from Vendor connection so it initiates a special authorization for this call the call will be billed to the account which receives it Thus the maximum call time duration is calculated based on A s current balance e Inthe authorization response PortaSIP is instructed to send the call to A s SIP phone 12027810003 4 e PortaSIP sends a call setup request to the SIP phone 5 e If the dialed number belongs to a SIP account with unified messaging services enabled and the account is not online at the moment or does not answer the call will be redirected to a voicemail system After the call is completed A is charged for it also costs are calculated for the incoming call according to the tariff associated with X Telecom s VoIP from Vendor connection 2000 2006 PortaOne Inc All rights reserved www portaone com 20 Porta SIP System Concepts Virtual SIP Servers On a single PortaSIP installation one physical server one license you can run multiple virtual PortaSIP instances each of them a separate server that can be used in a PortaBilling virtual environment The only thing required to create a new SIP instance on the SIP server side is adding an extra IP address IP alias and populating the configuration files Please contact the PortaOne support te
29. SIP UA and PSTN GW while for SIP UA lt gt SIP UA calls the RTP path depends on whether or not an RTP proxy is enabled If an RTP proxy is not enabled the RTP flows directly from one SIP UA to another Otherwise each RTP packet sent by one UA goes first to the machine running PortaSIP and is then resent from that machine to another SIP UA tried to register with the SIP server but my UA says registered even if my username or password are incorrect is there a security breach in PortaSIP Of course PortaSIP does not really allow unauthorized clients onto your network If the SIP UA tries to register using an incorrect username or password or with an account which is blocked registration will not succeed However UA will still receive registration confirmation and this is why you see registered in the UA But if you try to make an outgoing call it will be diverted to the media server where the appropriate message will be played e g This account does not exist or Account is blocked This allows SIP registration s troubleshooting to be greatly simplified Keep alive functionality does not work with my XXX brand SIP phone Your SIP phone must correctly respond to keep alive re INVITE requests If it does not support this functionality then it may either not reply at all to these requests or even worse assume that this is a new incoming call If PortaSIP detects that the SIP UA has not answered the fi
30. This number will be returned to the node which requested it Connection based outgoing translation rules If your vendor requires a special number format e g tech prefix it is possible to apply this rule to convert the number When billing returns a list of routes the phone numbers for routes for this connection will be converted This only works for a routing model in which the VoIP node e g PortaSIP requests billing for routing information If your gateway uses dial peers or an external gatekeeper for routing then you must configure number translation there Connection based translation rules When the call has been terminated to the vendor in a vendor specific format it will be reported to billing in this same format e g 7834 42021234567 Now it is necessary to convert this number to the proper format for billing 4202134567 which may be done using connection translation rules These rules will be applied to all calls which go through a given connection even those routed there using dial peers or other external tools Node based translation rules These serve the purpose of converting a number from a custom format used by the customer into billing s internal format during authorization depending on the gateway For example on a gateway in Prague Czech Republic there may be the translation rule strip leading 00 while on a 2000 2006 PortaOne Inc All rights reserved www portaone com 39 Porta SIP System
31. about the specific IP phone in the IP phone inventory including the device s MAC address with a specific profile assigned to it e The IP phone or in the case of a multi line device a port on the phone is assigned to a specific account in the billing Auto provisioning will only work if your IP phone knows the address of your provisioning server If you buy IP phones retail you will probably have to change the address of the provisioning server on every phone manually However if you place a large enough order with a specific vendor these settings can be pre configured by him so that you may deliver an IP phone directly to the end user without even unwrapping it IP Phone Inventory The IP phone directory allows you to keep track of IP devices SIP phones or adaptors which are distributed to your customers The MAC address parameter is essential for every IP phone which is to be automatically provisioned and so a corresponding entry must be created in the IP phone inventory 2000 2006 PortaOne Inc All rights reserved www portaone com 49 Porta SIP Setting up SIP Services 2 Setting up SIP Services Please refer to the PortaBilling100 Administrator Guide PDF file for detailed instructions on how to navigate and operate the web interface and detailed explanations of particular fields 2000 2006 PortaOne Inc All rights reserved www portaone com Porta SIP Setting up SIP Services Basic SIP service
32. and a percent sign for example 380 o From To Date the date range Click on the 10 min icon to limit the time interval to the last 10 minutes only Click Trace a Call The advantage of this method you may view all the call attempts including unsuccessful calls with disconnect reasons displayed Also you can see the billing history for a call For administrator convenience accounts CDRs may also be accessed from the Account Management window by clicking the CDR icon for an account 2000 2006 PortaOne Inc All rights reserved www portaone com 93 Porta SIP Setting up SIP Services Setting up auto provisioning of IP phones The following section will assist you in setting up automatic provisioning of IP phones by PortaSwitch so that every phone will automatically download all the required configuration parameters such as phone number or password from the provisioning server Checklist Print the following page and use it to check off the operations you have completed while performing system setup according to the instructions in this section Please make sure that you perform all of the operations all of the boxes must be checked otherwise the service will not work Operation Done General configuration Create a new IP phone profile for the required type of IP phone In the IP phone profile fill in all the required parameters e g SIP proxy address IP phone inventory Create a reco
33. by the vendor to retrieve a configuration from your provisioning server you need only connect the phone to the Internet Otherwise you must enter information regarding the provisioning server into the IP phone manually 1 Connect the Sipura device to the LAN and find the IP address assigned to it by DHCP connect an analog phone to phone port 1 and then dial on the phone so that you enter the configuration menu Dial 110 and listen to the IP address announced 2 On the PC connected to the same LAN as the Sipura device open a new web browser window and enter the URL http lt IP gt where lt IP gt is the IP address which was announced to you by the IVR 3 Click on Admin login then go to Advanced view and the Provisioning tab Address http 192 168 0 237 SIPURA technology inc Sipura Phone Adapter Configuration Info System User 1 User 2 Semin Login basic advanced http 192 168 0 237 admin SIPURA technology inc Info System SIP Regional Phone Line 1 Line 2 User 1 User 2 User Login basic advanced 2000 2006 PortaOne Inc All rights reserved www portaone com 1 00 Porta SIP Setting up SIP Services Address http 192 168 0 237 admin advanced v Go Links technology inc amp Sipura Phone Adapter Configuration Info System SIP Provisioning Regional Phone Line 1 Line 2 User 1 User 2 User login basie advanced
34. communicate with each other directly while in cases 1 and 3 it is possible to arrange things so that a media stream flows directly from one endpoint to another 2000 2006 PortaOne Inc All rights reserved www portaone com 4 5 Porta SIP System Concepts Calls between SIP phones 1 call is made from one SIP UA SIP phone to another SIP UA SIP phone with both phones on public IP addresses outside a NAT In this case the phones can communicate directly and no RTP proxying is required 2 call is made from one SIP UA SIP phone to another SIP UA SIP phone and at least one of the phones is on a private network behind a NAT Here an RTP proxy should be used to prevent no audio problems 3 call is made from one SIP UA SIP phone to another SIP UA SIP phone with both phones on the same private network behind the same NAT This scenario is likely to be encountered in a corporate environment where a hosted IP PBX service is provided In this case it is beneficial to enable both phones to communicate directly via their private IP addresses so that the voice traffic never leaves the LAN Calls between SIP phones and PSTN 1 A call is made from to a SIP phone on a public IP address from to a VoIP GW a VoIP GW is always assumed to be on a public IP address In this case the RTP stream may flow directly between the GW and SIP phone and no RTP proxying is required 2 A call is made from to a UA
35. gateway can place the outgoing calls and is able to communicate with the billing using RADIUS SIP server parameters Specify general parameters of the SIP server such as hostname You can also refer to the SIP server by its IP address however this method will require reconfiguration of each individual gateway if you change the IP address of your SIP server Sip ua aaa username proxy auth Sip server dns lt hostname of your SIP server gt NOTE Cisco GWs are currently unable to register to SIP servers using the REGISTER method or to perform proper authorization of an outgoing call using the INVITE method Therefore remote IP address authorization is performed by PortaSIP when it detects an incoming call from the Cisco gateway In order for this authorization to be successful the gateway should be registered among the PortaBilling nodes Dial peers Now you can create an SIP enabled outgoing dial peer dial peer voice 200 voip destination pattern T session protocol sipv2 2000 2006 PortaOne Inc All rights reserved www portaone com 1 05 Porta SIP How to session target sip server You probably will need an application on the incoming telephony dial peer to properly authenticate and authorize incoming calls configure my Cisco gateway for PSTN gt SIP service Obtain a PSTN2SIP application Create an application and a dial peer to process incoming PSTN calls Gall application voice pstnZsip flash
36. in exactly the same way as they used to do it on their PBX 9 for the outside line then 00 for the international dialing or O for domestic Clearly there is a need for the translation rule and there is one customer based translation rule Moreover to give the customer ability to manage his translation rule himself without the necessity to learn regular expressions there is a wizard which allows to construct the rule by just entering the main parameters such as international dialing prefix HR Edit Customer Fe TE ads Gl Save dose cons E accounts it Bios Customer Name EasyCall Ltd i Opening Balance 0 00000 USD Blocked o Balance 0 00000 USD Type Retail Address Info Maintenance Additional Info Payment Info User Interface Dialing Rules Notepad Abbreviated Number Length oO Dialing rules wizard gt bel save i Save amp Close Close Your country code Sample settings Your area code s North America WA 7 digit dialing Always dial the area code as a part of the number North America BC 10 digit dialing Europe Czech Rep always dial using the areacode Prefix for accessing the outside phone network Europe Czech Rep local and domestic dialing obsolete 3 Australia Sydney Prefix for domestic calls but outside of your area au ane International dialing prefix e g 011 00 0011 Emergency numbers e g 911 112 Exceptions e g 98
37. list o If the follow me order is Random then the list of phone numbers is shuffled o The maximum call duration is calculated for each follow me number based on the balance and rates for the called account A Oo The resulting list of routes is produced and sent back to PortaSIP 6 PortaSIP tries the first route 7 if the call is not connected within the timeout interval it moves to the next route 8 then to the next one 9 until either the call is put through or no more routes are left If such a call was completed to follow me number R two CDRs will appear in the system one for the call C gt A charged per the incoming rates for A and the other for C gt R charged per the outgoing rates for A If the call did not originate in the PST N network but rather from user B s SIP UA two CDRs will likewise be generated B will be charged for call B gt A while A will be charged for call B gt R Follow me service can be recursive Thus A can forward calls from his SIP phone to B s SIP phone and B can forward calls to his mobile phone number C Note that in the case of such a multi hop follow me A gt B gt C gt D gt PSTN number only two CDRs will be produced similar to a simple follow me a CDR for the caller billed to A A gt B a CDR for the forwarder outside the network 1 e the last SIP account in the follow me chain billed to D A gt PSTN 2000 2006 PortaOne Inc All rights rese
38. make this call on behalf of A since even if we know A s account ID we do not know A s password therefore such a call will be rejected In addition Cisco gateways currently do not support INVITE with authorization 2000 2006 PortaOne Inc All rights reserved www portaone com 1 9 Porta SIP System Concepts e PortaSIP receives the INVITE but without authorization information So the PortaSIP server performs authentication based on the IP address 5 6 Since this call is made from our trusted node gateway GW NY 01 the call is authorized e PortaSIP checks if the SIP user agent of the dialed number 12027810003 is registered at the time If yes a call setup request is sent 7 e If the dialed number belongs to an SIP account with unified messaging services enabled but this account is not online at the moment or does not answer the call will be redirected to a voicemail system e After the call is completed the B2BUA sends accounting information for the two VoIP call legs to the billing The gateway will also send accounting information about the answet Telephony and originate VoIP call legs The billing engine will combine this information since accounting from the SIP server allows us to recognize that the call was terminated directly to the SIP user agent and not to a vendor while accounting from the gateway will contain information as to which account should be billed for this call PSTN gt SIP vi
39. tag a4044d2fe886380800 To lt sip 380443333333 70 68 0 213 gt Call ID lefdc57b 80c3c2c 192 168 1 180 CSeq 101 ACK Max Forwards 70 Contact 758 lt sip 18667478647758 192 168 1 180 5060 gt User Agent Sipura SPA2000 2 0 13 g Content Length 0 SER receives a reply to the authorization request with a response to the challenge 21 Jul 23 37 26 GLOBAL ser RECEIVED message from 62 244 32 30 50563 INVITE sip 380443333333 70 68 0 213 SIP 2 0 Via SIP 2 0 UDP 192 168 1 180 5060 branch z9hG4bK 443b706f From 758 lt sip 18667478647758 70 68 0 213 gt tag a4044d2fe886380800 TO lt sip 380443333333 70 68 0 213 gt Call ID 1l1efdc57b 80c3c2c 192 168 1 180 CSeq 102 INVITE Max Forwards 70 Authorization Digest username 18667478647758 realm 70 68 0 213 nonce 5dbe8444a6063753 3 5c247b356641ca uri sip 380443333333 70 68 0 213 algorithm MD5 respon se 7f3a6offfcbl1 68b560a20937343e3ef0 Contact 758 lt s10 1866 4 78047 7580192 168 1 180 5060 gt Expires 240 User Agent Sipura SPA2000 2 0 13 g Content Length 420 Allow ACK BYE CANCEL INFO INVITE NOTIFY OPTIONS REFER Supported x sipura Content Type application sdp v 0 o 2284 2284 IN IP4 192 168 1 180 oo c IN IP4 192 168 1 180 t 0 0 m audio 16384 RTP AVP 18 0 2 4 8 96 97 98 100 101 a rtpmap 18 G729a 8000 a rtpmap 0 PCMU 8000 a rtpmap 2 G726 32 8000 a rtpmap 4 G723 8000 a rtpmap 8 PCMA 8000 a rtpmap 96 G726 40 80
40. the 1st Monthly 1 last day inclusive sent on the 1st of the following month Payment info tab o Credit limit if left empty then there is no credit limit for this customer o Balance Warning Threshold the customer can be notified by email when his balance is dangerously close to the credit limit and service will soon be blocked Here you can enter the value for such a warning threshold This can be entered Asa percentage e g 90 The warning will be sent when the customer s balance exceeds that percentage of his credit limit So if the credit limit is USD 1000 00 and the threshold is 90 a warning will be sent as soon as the balance exceeds USD 900 00 This is only applicable when the customer has a positive credit limit As an absolute value The warning will be sent as soon as the balance goes above the specified value User Interface tab o Time zone time zone in which customer will see his CDRs and also the time zone which will define his billing period For example if you choose America New_York here and the billing period is Monthly it means the billing period will start on the first day of the month 00 00 New York time o Web Interface Language language to be used on the customer self care web interface 4 Click ml Save amp Close Create Accounts 1 Go to the Customers screen the one containing the list of customers It should resemble the screenshot below i 4 Customer M
41. their SIP regardless of where they are at the moment In order to do so you will need to obtain a range of phone numbers from your telecom operator and make sure that calls made to these numbers on the PSTN network are routed to your gateway via the telephony interface e C wishes to call A He thus dials A s phone number since C is in the US he dials it using the North American format 2027810003 e This call is routed through the telecom network to gateway GW NY 01 When the incoming call arrives on the gateway 1 it starts a special TCL application PSTN2SIP to handle this call This application does several things O O Converts the phone number to the E 164 format so that 2027810003 become 12027810003 Performs authorization in the billing 2 whether A is allowed to receive incoming telephony calls from GW NY 01 and if you charge for incoming calls what is the maximum call time allowed based on A s current balance 3 One important point is that authorization should happen without a password check since the application does not know the valid password for the SIP account Starts outgoing call to 12027810003 Starts the timer once the call is established disconnecting the call when the maximum call duration is exceeded The gateway is configured such that it knows that calls to 1202781 numbers should be sent to the PortaSIP server thus it sends an INVITE to PortaSIP 4 NOTE The gateway cannot
42. to in the routing information e PortaSIP receives its own address as the route and sends a call to the SIP phone 2000 2006 PortaOne Inc All rights reserved www portaone com 9 4 Porta SIP System Concepts Case B SIP phones registered to different PortaSIP servers In this case routing information from PortaBilling will contain the address of the second PortaSIP server i e the one to which the called SIP phone is registered Thus the first PortaSIP server will send a call there and then the second PortaSIP server will send the call to the SIP phone e eo Porta 4 Billing Es Eo Billing Engine It may be asked why the first originating PortaSIP server does not send the call directly to the called SIP phone since the registration database contains its contact IP port information The answer is that if the called SIP phone is behind a NAT and most Internet users are behind a NAT these days only the server on which the SIP phone has opened a connection can send back a reply and this is the second PortaSIP server Note that although SIP signaling will travel via both SIP servers this is not the case with RTP voice traffic Depending on the NAT context of the call and the RTP proxy configuration PortaSwitch may either connect the RTP stream between the phones directly or use the RTP proxy on one of the SIP servers So even if two SIP servers are involved in this call this does not affect
43. your routing preferences E A X termination tariff tariff rates gt en add save close rar objets it Bix Effective From Destination mow Prefix Corcur Ceountey Country Routing Interval sec Price USD min Effective From Edit Destination k YYYY MM DD T Delete Description Route Category Preference Huntstop First Next First Hext HH Mi fn ie CANADA 2005 06 21 1604 Good 6 1 1 0 06000 0 06000 EI British Columbia 5 13 18 52 E 1 pote olan Default 5 1 1 0 05000 005000 2005 06 21 North America 13 18 18 UNITED KINGDOM _06 E 44 Cheap 3 1 1 0 01000 0 01009 2005 06 21 Proper 13 18 46 o Route category you can split your available routes into several categories such as Cheap Very good etc then create routing plans for your customers Use the Default route category for now o Preference routing priority for the specific destination 10 is the highest priority 0 is the lowest i e do not use destination for routing at all For now you can just set all of your vendor rates at preference 5 and the system will organize available routes according to cost LCR o Huntstop do not try any routes with a lower preference Managing rates offline NOTE Templates are available in PortaBilling a powerful tool for uploading rates from custom format data files However in this particular example we assume that you will ente
44. 00 a rtpmap 100 NSE 8000 a rtpmap 101 telephone event 8000 a fmtp 101 0 15 a ptime 30 a sendrecv a direction active a oldmediaip 192 168 1 180 In the rest of the log example we will skip the request transmission between SER and B2BUA request received by SER request sent to B2BUA request received by B2BUA since this would only duplicate the same information 21 Jul 23237226 GLOBAL ser SENDING message to 62 244 32 30 50563 PortaSIP requests digest authentication from the SIP UA providing a challenge SIP 2 0 401 Unauthorized 2000 2006 PortaOne Inc All rights reserved www portaone com 76 Porta SIP Setting up SIP Services Vias STP 2 0 UDE 192 168 1 180 5060 received 62 244 32 30 rport 50563 branch z 9hG4bK 6ad8150b Record Route lt sip 7 0 68 0 213 ftag a4044d2fe886380800 1r gt From 758 lt sip 18667478647758 70 68 0 213 gt tag a4044d2fe886380800 To lt sip 380443333333 070 68 0 213 gt Call ID 1l1efdc57b 80c3c2c 192 168 1 180 CSeq 101 INVITE Server Sippy WWW Authenticate Digest realm 7 0 68 0 213 nonce 5dbe844446063 531f35c247 b356641ca 21 Jul 23337 26 GLOBAL ser RECEIVED message from 62 244 32 30 50563 SIP UA acknowledges that it has received an authorization request ACKs will be Skipped in the rest of the document ACK sip 380443333333 70 68 0 213 SIP 2 0 Via SIP 2 0 UDP 192 168 1 180 5060 branch z9hG4bK 6ad8150b From 758 lt sip 18667478647758 70 68 0 213 gt
45. 00 a rtpmap 97 G726 24 8000 a rtpmap 98 G726 16 8000 a rtpmap 100 NSE 8000 a rtpmap 101 telephone event 8000 a fmtp 101 0 15 a ptime 30 2000 2006 PortaOne Inc All rights reserved www portaone com 77 Porta SIP Setting up SIP Services a sendrecv SER nepecpimaet Banpoc Ha b2bua B NaKxetTe xexaeTC4 MOHMOMKauna aoGagienne PortaBilling Notify NAT cm Hnxe n BameHa c line B SDP Ha PeaJBHBM aupec gna ero s3HaHnna B b2bua 21 Jul 23 37 26 GLOBAL ser SENDING message to 70 68 0 213 5061 This request is resent to B2BUA with several modifications in particular a PortaBilling Notify NAT flag is added to inform B2BUA of the NAT status of the device INVITE Ssip 380443333233070 68 0 213 5061 SIP 2 0 Record Route lt sip 70 68 0 213 ftag a4044d2fe886380800 1r gt Via SIP 2 0 UDP 70 68 0 213 branch z9hG4bK2e23 c0167 c9b6de2bee2cheec845eac8F0d6 0 Viat SIP 2 0 7UDE 192 168 1 180 5060 rport 50563 received 62 244 32 30 branch z9hG4bK 443b706f From 758 lt sip 18667478647758 70 68 0 213 gt tag a4044d2fe886380800 To lt sip 380443333333070 68 0 213 gt Call ID lefdc57b 80c3c2c 192 168 1 180 CSeq 102 INVITE Max Forwards 16 Authorization Digest username 18667478647758 realm 70 68 0 213 nonce 5dbe8444a6063753 3 5c247b356641ca uri sip 380443333333 70 68 0 213 algorithm MD5 respon se 7f3a6fffch1lf68b560a20937343e3ef0 Contact 758 lt sip 18667478647758062 244 32 30 50563 gt
46. 2 priority 0004 cost 0 1606 There are three possible routes sorted according to preference and cost 2000 2006 PortaOne Inc All rights reserved www portaone com 87 Porta SIP Setting up SIP Services Jul 21 22 37 26 Logging in account 18667478647758 147059 to 9C25D01D 9ED3C282 83BAC556 8CE06004 Jul 21 22 37 26 Authentication acknowledge response Authorization response is sent to PortaSIP Cisco AVPair h323 1vr 1n PortaBilling Routing gt 380443333333 195 234 212 1 expires 300 credit time 1 Cisco AVPair h323 ivr in PortaBilling Routing gt 380443333333 66 96 26 134 5061 expires 300 credit time 1 Cisco AVPair h323 ivr in PortaBilling Routing gt 380443333333 67 105 130 102 auth expires 300 credit time 1 h323 billing model 0 h323 ivr in Tariff Porta SIP h323 ivr in PortaBilling CompleteNumber 3604433353333 h323 return code 13 h323 currency USD h323 preferred lang en Accounting for the failed outgoing call leg arrives Jul 21 22 37 59 Processing request BE ver1 218 2 2 pid24729 NAS IP Address 70 68 0 213 User Name 18667478647758 Called Station Id 390443333333 Calling Station Id 18667478647758 Acct Status Type Stop h323 call origin originate h323 call type VoIP h323 setup time h323 connect time h323 disconnect time OG 37 26 000 GMT Fri Jul 22 20057 06 37 58 000 GMT Fri Jul 22 20057 06 37 58 000 GMT F
47. 34 gt tag a35c6f23a59ed2101cd78ddf276cdf9e To lt sip 380443333333066 96 26 134 gt Call ID lefdc57b 80c3c2c 192 168 1 180 CSeq 200 INVITE Contact Anonymous lt sip 70 68 0 213 5061 gt Expires 300 User Agent Sippy cisco GUID 2619723805 2664678018 2210055510 2363514890 h323 conf id 2619723805 2664678018 2210055510 2363514890 Content Length 466 Content Type application sdp v 0 o 2284 2284 IN IP4 192 168 1 180 0 0 audio 35134 RTP AVP 18 0 2 4 8 96 97 98 100 101 c IN IP4 70 68 0 213 a rtpmap 18 G729a 8000 a rtpmap 0 PCMU 8000 a rtpmap 2 G726 32 8000 a rtpmap 4 G723 8000 a rtpmap 8 PCMA 8000 a rtpmap 96 G726 40 8000 a rtpmap 97 G726 24 8000 a rtpmap 98 G726 16 8000 a rtpmap 100 NSE 8000 a rtpmap 101 telephone event 8000 a fmtp 101 0 15 a ptime 30 a sendrecv a direction active a oldmediaip 192 168 1 180 S t m This gateway is available so we get a reply that it has started to establish the call 21 Jul 23 37 59 GLOBAL b2bua RECEIVED message from 66 96 26 134 5061 SIP 2 0 100 Trying Via SIP 2 0 UDP 70 68 0 213 5061 branch z 9hG4bKd7 9ceb45f 97 3bef2a3747381d8a70392 rport From 758 lt sip 1666 7476647758 66 96 26 134 gt tag a35c6f23a59ed2101cd78ddf276cdf9e fos lt sip 380443333333 66 96 26 134 gt tag ffff4f000fffff10ff00000255ffff4b Call ID 1l1efdc57b 80c3c2c 192 168 1 180 CSeq 200 INVITE Contact lt sip 380443333333 66 96 26 134 5061 user phone gt Server MERA MS
48. 356641ca uri sip 70 68 0 213 5061 algorithm MD5 response l1dle elfbf0ae1aa58450a413dfb7f6bf Contact 758 lt s1ip 1860674780417580192 168 1 180 5060 gt User Agent Sipura SPA2000 2 0 13 g Content Length 0 RTP stream from the caller is established the call is in progress 21 Jul 23 38 07 1efdc57b 80c3c2c 192 168 1 180 rtpproxy caller s address filled in 62 244 32 30 50569 RTP One of the parties hangs up the call termination process is started 21 Jul 23 38 24 GLOBAL b2bua RECEIVED message from 66 96 26 134 5061 BYE sip 70 68 0 213 5061 SIP 2 0 Via SIP 2 0 UDP 66 96 26 134 5061 From lt sip 380443333333 66 96 26 134 gt tag ffff4f000fffff10ff00000255ffff4b To 758 lt sip 18667478647758 66 96 26 134 gt tag a35c6f23a59ed2101cd78ddf276cdf9e Call ID l1efdc57b 80c3c2c 192 168 1 180 CSeq 201 BYE Max Forwards 10 User Agent MERA MSIP v 1 0 1 Reason Q 850 cause 16 text Normal call clearing Content Length 0 B2BUA sends stop accounting to the billing for the outgoing call leg 21 Jul 23 38 24 1efde5 b 80c3c2c0192 168 1 180 b2bua sending Acct Stop Originate User Name 1866 7478647758 Calling Station Id 1866 7478647758 Called Station Id 380443333333 h323 call origin originate h323 call type VoIP h323 conf id 9C25D01D 9ED3C282 83BAC556 8CEQ600A call id lefdc57b 80c3c2c 192 168 1 180 l1efdc57b 80c3c2c 192 168 1 180 666 96 26 134
49. 61 branch z9hG4bK4de91149a7529218ce70 b920 9be2ef9 rport 5 061 Server Sipura SPA2000 2 0 13 g Content Length 0 The call is finished 6 Browse information in the PortaBilling log file To do so you can either e Login to the PortaBilling master server and type less vart Log porta billing log e Find this call using Trace call on the PortaBilling web interface view the call details press Fe icon in the leftmost column and then click the 1 View log button in the tool bar PortaBilling receives the authorization request Jul 21 22 37 26 Processing request BE ver1 218 2 2 pid24729 NAS IP Address 70 68 0 213 User Name 18667478647758 Called Station Id 380443333333 A 18667478647758 9C25D01D 9ED3C282 83BAC556 8CEO600A l1efdc57b 80c3c2c 192 168 1 180 Calling Station Id h323 conf id Call id I A Digest Attributes Realm 70 68 0 213 Digest Attributes Nonce 5dbe8444a6063753 35c247b356641ca Digest Attributes Method INVITE Digest Attributes URI sip 380443333333 7 70 68 0 213 Digest Attributes Algorithm MD5 Digest Attributes User Name 18667478647758 Digest Response 7 f3a6fffch1lf68b560a20937343e3ef0 h323 remote address None h323 session protocol sipyv2Z h323 ivr out t PortaBilling Routing SIP h323 ivr out PortaBilling Notify NAT NAS Port Id 50607
50. 8 G729a 8000 2 G 26 32 8000 96 G726 40 8000 97 G726 24 8000 98 G726 16 8000 101 telephone event 8000 a oldmediaip 192 168 1 180 After several attempts no reply is received from the remote gateway proxy This route is considered to have failed and an accounting record is sent to the billing 21 Jul 23 37 59 1lefdc57b 80c3c2c 192 168 1 180 b2bua Originate User Name Calling Station Id Called Station id h323 call origin h323 call type h323 conf id call id Acct Session Id h323 remote address h323 session protocol h323 setup time h323 disconnect time h323 connect time Acct Session Time h323 disconnect cause Acct Status Type sending Acct Stop 18667478647758 18667478647758 380443333333 originate VOLE 9C25D01D 9SED3C282 83BAC556 8CEHOG00A l1efdc57b 80c3c2c 192 168 1 180 l1efdc57b 80c3c2c 192 168 1 180 195 234 212 1 sipv2 06 37 26 000 GMT Fri Jul 22 2005 06 37 58 000 GMT Fri Jul 22 2005 06 37 58 000 GMT Fri Jul 22 2005 V0 66 Stop RTP proxy session is closed as well 21 Jul 23 37 59 GLOBAL rtpproxy received command D lefdc57b 80c3c2c 192 168 1 180 e97ded4c2b244b0cb4007cedb742 088 543650df2972e81b25e307d ed2a2e04 21 Jul 23 37 59 1efdc57b 80c3c2c 192 168 1 180 rtpproxy forcefully deleting session 0 on ports 35132 0 21 Jul 23 37 59 1efdc57b 80c3c2c 192 168 1 180 rtpproxy O in from caller 21 Jul 23 37 59 1efdc57b 80c3c2c 1
51. 92 168 1 180 rtpproxy O in from caller 21 Jul 23 37 59 1efdc57b 80c3c2c 192 168 1 180 rtpproxy in from callee in from callee RTP stats 0 O relayed 0 dropped RTCP stats 0 O relayed 0 dropped session on ports 35132 0 is cleaned up 21 Jul 23 37 59 GLOBAL rtpproxy sending reply 0 PortaSIP attempts to send the call via the second route 21 Jul 23237 59 GLOBAL rtpproxy received command U lefdc57b 60630200192 168 1 180 62 244 32 30 16384 813da13c69239f8d427bbd3b78c601ee 5d4911c473909f6d59b54af2f14f12986 2000 2006 PortaOne Inc All rights reserved www portaone com Porta SIP Setting up SIP Services 21 Jul 23 37 59 GLOBAL rtpproxy new session lefdc57b 80c3c2c 192 168 1 180 tag 813dal3c69239fFf8d427bbd3b78c60lee medianum 0 requested type strong 21 Jul 23 37 59 1efdc57b 80c3c2c 192 168 1 180 rtpproxy new session on a port 35134 created tag 813dal13c69239f8d427bbd3b78c601lee 21 Jul 23 37 59 1efdc57b 80c3c2c 192 168 1 180 rtpproxy pre tilling caller s address with 62 244 32 30 16384 21 Jul 23 37 59 GLOBAL rtpproxy sending reply 35134 70 68 0 213 INVITE is sent to the next gateway proxy in the route list 21 Jul 23 37 59 GLOBAL b2bua SENDING message to 66 96 26 134 5061 INVITE sip 380443333333 66 96 26 134 5061 SIP 2 0 Vlast SIP 270 UDP 70 68 0 213 5061 branch z9hG4bKd79ceb45f973bef2a3747381d8a70392 rport Max Forwards 70 From 758 lt sip 18667478647758 66 96 26 1
52. 9ceb45 f973bef2a3747381dqd8a70392 rport From 758 lt sip 18667478647758 66 96 26 134 gt tag a35c6f23a59ed2101cd78ddf276cdf9e To lt sip 380443333333 66 96 26 134 gt tag ffff4f000fffff10ff00000255ffff4b Call ID 1l1efdc57b 80c3c2c 192 168 1 180 CSeq 200 INVITE Contact lt sip 380443333333 66 96 26 134 5061 user phone gt Server MERA MSIP v 1 0 1 Content Type application sdp Content Length 264 v 0 o 1122014231 1122014231 IN IP4 66 96 26 134 c IN IP4 66 96 26 134 t 0 0 m audio 20620 RTP AVP 18 0 4 8 101 a rtpmap 18 G729 8000 a rtpmap 0 PCMU 8000 a rtpmap 4 G723 8000 a rtpmap 8 PCMA 8000 a rtpmap 101 telephone event 8000 a fmtp 101 0 15 Start accounting for the outgoing originate VolP call leg is sent to the billing 21 Jul 23 38 07 1efdc57b 80c3c2c 192 168 1 180 b2bua sending Acct Start Originate User Name 18667478647758 Calling Station Id 1866 7478647758 Called Station Id 380443333333 h323 call origin originate h323 call type VoIP h323 conf id 9C25D01D 9ED3C282 83BAC556 8CEO600A 2000 2006 PortaOne Inc All rights reserved www portaone com 99 Porta SIP Setting up SIP Services call id lefdc57b 80c3c2c 192 168 1 180 Acct Session Id Jefdc57b 80c3c2cQ 192 168 1 180 h323 remote address 66 96 26 134 h323 session protocol sipyz h323 setup time 06 37 59 000 GMT Fri Jul 22 2005 h323 connect time 06 38 07 000 GMT Fr
53. D 00 00 HH Mi Pages 1 2 3 4 5 6 7 8 9 10 11 gt gt Total 19 View CLiani CLD dnis Country Description 7 greoa aes seach Amount Account Customer Vendor gl E E 40977549630 16047202030 ho Ne 0 00 0 USD 56 78 90 1 Patel SuperNet Awa E E 15164433617 16045722686 Cane ae 0 00 0 USD 56 78 901 Us SuperNet Temporary fare E E 18315758558 16040620680 pees cee a ee 10 46 sai 56 78 90 1 ri Fu Se E E 71541627115 16043809353 une See eg 0 00 OUSD xtelecom XTelecom ne RE E E o2430543246 16042354467 sure EE 0 00 0 USD 56 78 90 ne Be ees eee E E 71113435558 16044514881 Pree Spree lanai 6 14 SEE telecom X Telecom nue RTE E E 53885854242 16043410791 Piss cee eee 0 00 0 USD 56 78 901 rapes nes User busy ee ee OUSO Cea E E 18165706423 16047142557 AA eg ee 2 58 MPE telecom X Telecom nR E A 2 E 40115171017 16042595019 sable Ne 1 25 D 100003 ae SuperNet ES W E 27775191217 16045852627 pe es eee 0 00 OUSD 100003 eae yes lt a List of possible Disconnect reasons Normal completed call E Normal uncompleted call il Call progress code i Calling side error Bi Called side error EF Network error E Inthe Helpdesk section of Admin Index choose Trace a Call Fill in the check phone number form o h323 conf id if you need to trace a specific call enter h323 conf id here otherwise this leave empty o Destination the phone number you are looking for or a destination pattern first digits
54. IP v 1 0 1 Content Length 0 Ringback is received from the remote gateway 2000 2006 PortaOne Inc All rights reserved www portaone com 81 Porta SIP Setting up SIP Services 21 Jul 23 38 00 GLOBAL b2bua RECEIVED message from 66 96 26 134 5061 SIP 2 0 180 Ringing Vias SIP 2 0 UDP 70 68 0 213 5061 branch z9hG4bKd79ceb45 f973bef2a3747381dqd8a70392 rport From 758 lt sip 18667478647758 66 96 26 134 gt tag a35c6f23a59ed2101cd78ddf276cdf9e To lt sip 380443333333 66 96 26 134 gt tag ffff4f000fffff10ff00000255ffff4b Call ID l1efdc57b 80c3c2c 192 168 1 180 CSeq 200 INVITE Contact lt sip 380443333333 66 96 26 134 5061 user phone gt Server MERA MSIP v 1 0 1 Content Length 0 Ringback is transferred to the SIP UA so the user on the SIP phone will hear ringing 21 Jul 23 38 00 GLOBAL ser SENDING message to 62 244 32 30 50563 SIP 2 0 180 Ringing Via SIP 2 0 UDP 192 168 1 180 5060 received 62 244 32 30 rport 50563 branch z 9hG4bK 443b706f Record Route lt sip 7 0 68 0 213 ftag a4044d2fe886380800 1r gt From 758 lt sip 18667478647758 70 68 0 213 gt tag a4044d2fe886380800 To lt sip 380443333333 70 68 0 213 gt tag 54cce4d15b8 6cbhbe9dec50522083bflc Call ID 1l1efdc57b 80c3c2c 192 168 1 180 CSeq 102 INVITE Server Sippy The called party answers the call 21 Jul 23 38 07 GLOBAL b2bua RECEIVED message from 66 96 26 134 5061 SIP 2 0 200 OK Via SIP 2 0 UDP 70 68 0 213 5061 branch z9hG4bKd7
55. ITE S302 33044 3333333070 002 02219 Sie 2 0 Via SIP 2 0 UDP 192 168 1 180 5060 branch z 9hG4bK 6ad8150b From 758 lt sip 18667478647758 70 68 0 213 gt tag a4044d2fe886380800 To lt sip 380443333333 70 68 0 213 gt Call ID 1l1efdc57b 80c3c2c 192 168 1 180 CSeq 101 INVITE Max Forwards 70 Contact 758 lt sip 18667478647758 192 168 1 180 5060 gt Expires 240 User Agent Sipura SPA2000 2 0 13 g Content Length 420 Allow ACK BYE CANCEL INFO INVITE NOTIFY OPTIONS Supported x sipura Content Type application sdp v 0 o 2284 2284 IN IP4 192 168 1 180 s 2000 2006 PortaOne Inc All rights reserved www portaone com REFER 14 Porta SIP Setting up SIP Services c IN IP4 192 168 1 180 t 0 0 m audio 16384 RTP AVP 18 0 2 4 8 96 97 98 100 101 a rtpmap 18 G729a 8000 a rtpmap 0 PCMU 8000 a rtpmap 2 G726 32 8000 a rtpmap 4 G723 8000 a rtpmap 8 PCMA 8000 a rtpmap 96 G726 40 8000 a rtpmap 97 G726 24 8000 a rtpmap 98 G726 16 8000 a rtpmap 100 NSE 8000 a rtpmap 101 telephone event 8000 a fmtp 101 0 15 a ptime 30 a sendrecv 21 Jul 23 37 26 1lefdc57b 80c3c2c 192 168 1 180 ser processing INVITE received from 62 244 32 30 SIP user agent is informed that his request is being processed 21 Jul 23 37 26 GLOBAL ser SENDING message to 62 244 32 30 50563 SIP 2 0 100 trying your call is important to us Via SIP 2 0 UDP 192 168 1 180 5060 branch z9hG4bK 6ad8150b rport 50563 received
56. Jul 21 22 37 26 This call belongs to the environment 1 pb Jul 21 22 37 26 h323 conf 1d 9025D01D SED3C282 83BAC556 8CE0600A T Call id lefdce57b 80c36c260192 168 1 180 1 2000 2006 PortaOne Inc All rights reserved www portaone com 95 Porta SIP Setting up SIP Services Jul 21 22 37 26 H323 SIP call use h323 conf id but remember call id Jul 21 22 37 26 Trying to match connection for call Jul 21 22 37 26 Looking for a connection VoIP answer Jul 21 22 37 26 No original CLD using CLD from the request Jul 21 22 37 26 VoIP matching by the node IP 70 68 0 2137 remote ip 18667478647758 and CLD 380443333333 Jul 21 22 37 26 Unknown node IP or no VoIP from vendor connections for this node Jul 21 22 37 26 PrepareNexecute AccountAuth Jul 21 22 37 26 Found Account 18667478647758 147059 credit balance 0 55021 limit none of customer PortaOne 2 balance 9996029 96819 limit 0 00000 Account information is located in the database Jul 21 22 37 26 Account 18667478647758 is not logged in yet Jul 21 22 37 26 Verify password by Digest Response attributes Password verification is successful Jul 21 22 37 26 Applying customer dialing translation rule on CLD Jul 21 22 37 26 Translation s 011 defined 1 2 s 2 e ip 011 applied 380443333333 unchanged The customer number translation rule is applied but the number is unchanged Jul 21 22 37 26 PrepareNexecute Accoun
57. Jul 21 22 37 59 Connection to vendor not found on net call leg Jul 21 22 37 59 Accounting response Jul 21 22 37 59 Done 2000 2006 PortaOne Inc All rights reserved www portaone com 09 Porta SIP Setting up SIP Services The call is established start accounting arrives Jul 21 22 38 07 Processing request BE verl 218 2 2 pid24 29 NAS IP Address 70 68 0 213 User Name 18667478647758 Called Station Id 380443333333 Calling Station Id 18667478647758 Acct Status Type Start h323 call origin originate h323 call type VoIP 06 37 59 000 GMT Fri Jul 22 20057 06 38 07 000 GMT Fri Jul 22 2005 9C25D01D 9ED3C282 83BAC556 8CE06004A l1efdc57b 80c3c2c 192 168 1 180 l1efdc57b 80c3c2c 192 168 1 180 h323 setup time h323 connect time h323 conf idqd call id Acct Session Id Acct Delay Time 0 h323 remote address 66 96 26 1347 h323 session protocol sipv2 NAS Port Id pee Oho thd Exec Program Log Jul 21 22 38 07 This call belongs to the environment 1 pb Jul 21 22 38 07 h323 conf id 9C25D01D 9ED3C282 83BAC556 8CE0600A 1 call id lefdc57b 80c3c2c 192 168 1 180 1 Jul 21 22 38 07 Found a call in cache with such id Jul 21 22 38 07 Copied account 18667478647758 147059 credit balance 0 55021 limit none of customer PortaOne 2 balance 9996029 96819 1limit 0 00000 from 70 68 0 213 into the current request Jul 21 22 38 07 F
58. Mobile 30 6 0 17 0 17 30 6 0 15 0 15 420601 30 6 0 17 0 17 30 6 0 15 0 15 a Save the file in Excel You will probably get a warning from Excel that yout file ay contain features that are not compatible with CSV Comma delimited Ignore this and choose Yes to retain the CSV format 10 Close the file in Excel If you performed step 6 then disregard the message Do you want to save the changes you made since this arises only because your format is not the default Excel XLS format 2000 2006 PortaOne Inc All rights reserved www portaone com 60 Porta SIP 11 12 13 14 Setting up SIP Services Go back to the PortaBilling web interface and then go to the Tariff screen Click on the Upload button Either enter the name of your file manually or click Browse and choose the file Click Save amp Close You should return to the Tariff screen where a message will tell you about the status of the import Also you will receive an email confirmation of the tariff upload If any operations have failed you will receive whatever data was not uploaded as an attachment so you can try to import it later You can verify your work using the Edit Rates feature After you have done so go to the Main menu by clicking on the Home icon Create All Required Tariffs create K Repeat the Create Tariff and Enter Rates steps after which you will e A tariff for each account billing scheme For
59. NAS IP Address the IP address of the SIP server o Auth Translation rule Leave this blank you can use customer based translation rules later to allow your customers to dial a number in their own numbering format o Manufacturer select PortaOne o Type VoIP node type select PortaSIP o Radius Client check this since PortaSIP will need to communicate with the billing o Radius Key enter the radius shared secret here this must be the same key which you entered during the PortaSIP installation o Radius Source IP see the Node ID NAS IP address and Radius source IP section in PortaBilling100 User Guide for more information Unless your PortaSIP server uses multiple network interfaces the value here should be the same as the NAS IP Address 4 Click el Save amp Close 5 Repeat steps 2 4 for any additional gateways you may have Use VOIP GW as the node type NOTE There is some propagation delay between the database and the Radius server configuration file however it is no more than 15 minutes Create Tariff The tariff is a single price list for calling services or for your termination costs A tariff combines conditions which are applicable for every call regardless of the called destination per destination rates E 4 Tariff Management gt Gadd close i tego too Search z Currency Type Description Rates Delete lt 2000 2006 PortaOne Inc All rights r
60. P to vendor connection with remote IP identical to the SIP UA for this vendor using tariff C Account provisioning Create a retail customer who will use the SIP service Create several accounts for this customer with account ID identical to the SIP phone number Testing Program the parameters phone password SIP server address into the SIP phone and make a test call 2000 2006 PortaOne Inc All rights reserved www portaone com 51 Porta SIP Setting up SIP Services Initial Configuration of PortaBilling The following steps are normally performed only once after the system is installed Proceed as follows Visit Company Info on the main menu Enter information about your company and set up your base currency Naturally this does not limit your operations to this currency only However on cost revenue reports and the like different currencies will be converted to the one you specify here NOTE Once you set up a base currency it cannot be changed If you make a mistake you will have to start with a new PortaBilling environment From the main menu choose Users and create login entries for users who will be working with the system It is not recommended that the default PortaBilling root user pb root be used for any operations other than initial setup Make sure you are able to login as the newly created user and change the password for the pb root user If you plan to do billing in multiple
61. PortaBilling However this carrier requires the number to be in US dialing format so the international number must be prefixed by 011 An outgoing translation rule s 011 to carrier ABC has been defined and is now applied to the phone number with the result 01142021234567 Note that there may be several carriers who can terminate this call each with its own numbering 2000 2006 PortaOne Inc All rights reserved www portaone com 39 Porta SIP System Concepts format In such a case there will be several alternative routes with different phone numbers 4 PortaSIP attempts to establish a connection to remote gateway 1 2 3 4 using phone number 01142021234567 5 After the call is completed PortaSIP sends an accounting request to PortaBilling stating that a call to remote gateway 1 2 3 4 has just been completed PortaBilling finds a connection to vendor ABC with remote IP address 1 2 3 4 and applies the translation rule s 011 for this connection in order to convert the number from the vendor specific format into your billing format Thus 011 is removed from 01142021234567 and the number becomes 42021234567 PortaBilling searches for the vendor and customer rates for this number and produces the CDRs CLI translation rules off net calls CLI AND is the calling party number typically programmed on SIP phones However due to the reasons described above this number must be represented in a specific format dependin
62. PortaSIP User Guide Porta SIP PortaSIP User Guide Copyright notice amp disclaimers Copyright 2000 2006 PortaOne Inc All rights reserved PortaSIP User Guide December 2005 V 1 11 16 Please address your comments and suggestions to Sales Department PortaOne Inc Suite 400 2963 Glen Drive Coquitlam BC V3B 2P7 Canada Changes may be made periodically to the information in this publication Such changes will be incorporated in new editions of the guide The software described in this document is furnished under a license agreement and may be used or copied only in accordance with the terms thereof It is against the law to copy the software on any other medium except as specifically provided in the license agreement The licensee may make one copy of the software for backup purposes No part of this publication may be reproduced stored in a retrieval system or transmitted in any form or by any means electronic mechanical photocopied recorded or otherwise without the prior written permission of PortaOne Inc The software license and limited warranty for the accompanying products are set forth in the information packet supplied with the product and are incorporated herein by this reference If you cannot locate the software license contact your PortaOne representative for a copy All product names mentioned in this manual are for identification purposes only and are either trademarks or registered trademarks of
63. SIP server sends an authorization request to the billing 2 e Billing performs several operations O Checks that such an account exists that it is not blocked expired that the supplied password is correct that the account is allowed to use SIP services etc o Performs a dialed number translation according to the customer dialing rules or abbreviated dialing table 121 is converted to 12027810009 O Checks if A is actually allowed to call that number and what is the maximum allowed call duration O Checks whether the dialed number is one of our SIP accounts if it is currently registered and what is the NAT status of both SIP phones Based on the results of the above operations billing sends an authorization response to the SIP server 3 e The SIP server checks its registration database to find the actual contact address of the SIP user agent with that number 2000 2006 PortaOne Inc All rights reserved www portaone com 1 9 Porta SIP System Concepts The SIP server sends an INVITE to the SIP user agent for user B 4 If one of the SIP phones is behind NAT the SIP server will be instructed by the billing to send a voice stream via the RTP proxy Otherwise the SIP server may allow A and B s user agents to talk directly to each other When the call is finished the SIP server sends accounting information to the billing The called party is not online Porta M Billing Porta UM Unified Messagin
64. TP proxy policy for SIP PSTN calls RTPPRemotePolicy is a policy which defines when an RTP proxy will be engaged for calls going to third party vendors for PSTN termination or which come from PSTN and are delivered to SIP phones 1 direct do not use RTP proxy 2 nat use RTP proxy if the calling UA is behind NAT 3 all always use RTP Proxy By default the RTP policy for SIP PSTN calls is set to direct thus it is assumed that the remote gateway supports symmetric NAT traversal Comedia extensions 2000 2006 PortaOne Inc All rights reserved www portaone com 1 20 Porta SIP Administration FAQ Starting Stopping PortaSIP Services If you need to stop all PortaSIP services then execute the following command S sudo usr local erc rc d sip sh stop This will properly terminate all components To start PortaSIP use the following command S sudo usr local erc rc d sip sh start NOTE Please always make sure that you have stopped services as described above before trying to start them again since trying to start services when they are already running may render the service inoperable 2000 2006 PortaOne Inc All rights reserved www portaone com 1 21 Porta SIP Appendices 5 Appendices 2000 2006 PortaOne Inc All rights reserved www portaone com 1 99 Porta SIP Appendices APPENDIX A Tested Routers and NAT Software Commodity routers and NAT software bundled with popular operating sys
65. Timer 0 Polarity 0 ConnectMode 0xe0400 AuthMethod 0 TimeZone SEE CISCO ATA 186 DOCUMENTATION FOR ENTERING CORRECT ALUE NTPIP 192 43 244 18 AltNTPIP 131 188 3 222 DNS1IP 0 0 0 0 DNS2IP 0 0 0 0 UDPTOS Oxb8 SigTimer 0x64 OpFlags 0x62 VLANSettings 0x2b NPrintf 0 0 0 0 TraceFlags 0 The manufacturer s default values are assumed for all options not listed here 2000 2006 PortaOne Inc All rights reserved www portaone com 1 9 4 Porta SIP Appendices APPENDIX D Client s Sipura Configuration for PortaSIP 1 First you need to know the SPA IP address Via a touchtone telephone attached to the phone port on the SPA press the star key four times Then type 110 and the IP address will be announced 2 Runa Web browser application on the same network as the SPA Open a session in the SPA by typing http lt spa ip address gt admin advanced 3 Choose the specific phone port click on Line 1 Line 2 or another tab 4 Provide values for the required parameters which include a in Proxy and Registration i Proxy PortaSIP address or hostname ii Register yes b in the Subscriber information part i Display Name your identification e g John Doe this will be seen by the called party ii User ID SIP account ID ii Password VoIP password for your SIP account iv Use Auth ID no 5 Submit all the changes and update the SPA configuration 2000 2006 Port
66. XX000 as shown on the picture so every call will be rounded to the equal cent amount o Formula Default rating formula which will be applied to every rate created in the tariff If you leave this empty the old style rating will be used O Short Description a short tariff description This will be shown in the rate lookup on the admin interface and the self care pages for your accounts and customers o Description an extended tariff description 4 Click lel Save 5 Repeat steps 1 4 until you have entered all of the tariffs You will need at least two tariffs one which you will use to charge your customers and another which describes your termination costs Make sure you choose Routing in the Type select menu when creating tariffs for your vendors Enter Rates Rates are per destination prices Please refer to the System Concepts chapter for more details on billing parameters Managing rates online Managing rates online is very convenient for maintaining existing rate tables as well as for reference purposes For new price lists or for major updates an offline method is better GG M SIP Phone subscribers tariff rates fe E aaa te Save cle Stal Out Bis Effective From Destination Now E Prefix Group Core Country Interval sec Price USD min Effective From Edit Destination Ne Les A YYYY MM DD T delete Description First Hext First Next je HH24 MI SS Vv
67. a VoIP DID provider In the previous section we discussed traditional PSTN gt SIP service when a call is delivered to your gateway via E1 T1 lines and then forwarded to a SIP phone Unfortunately this service scheme assumes direct interconnection with the telco that owns DID numbers Establishing such direct interconnections with every telco from which you would like to get phone numbers can be problematic e g if you want to give your customers the ability to choose a phone number from any European country you will need many gateways in different places Fortunately however there are more and more companies which offer incoming DID service 1 e they already have an interconnection with a specific telecom operator and so can forward incoming calls on these numbers to you via IP Thus no extra investment is required to provide phone numbers from a certain country or area except signing a contract with such a DID consolidator 2000 2006 PortaOne Inc All rights reserved www portaone com 1 g Porta SIP System Concepts X Telecom Vendor Porta 4 Billing Phone C SIP phone A e C wishes to call A on his German phone number He thus dials A s phone number since C is in the US he dials it using the North American format 0114929876543 e The call is routed through the telecom network to the gateway of DID consolidator X Telecom 1 e X Telecom in turn forwards this call to your PortaSIP server 2 e
68. aOne Inc All rights reserved www portaone com 125 Porta SIP Appendices User Login basic advanced 2000 2006 PortaOne Inc All rights reserved www portaone com 1 26 Porta SIP Network Settings SIP TOS DiffSery Value RTP TOS DiffSery Value SIP Debug Option Call Feature Settings Blind Attn xXfer Enable xfer When Hangup Conf Proxy and Registration Proxy 216 231 44 168 Outbound Proxy Register Register Expires Use DNS SRY Proxy Fallback Intyl Subscriber Information eR Ro oo oe Mini Certificate SRTP Private Key Supplementary Service Subscription Call Waiting Serv yes Block ANC Serv yes Cfwd All Serv yes v Cfwd No Ans Serv yes M Cfwd Last Serv yes M Accept Last Serv yes M CID Serv yes M Call Return Serv yes M hree Way Call Serv yes M Attn Transfer Serv yes v Network Jitter Level SIP 100REL Enable 4uth Resync Reboot MOH Server Use Outbound Proxy Use OB Proxy In Dialog Make Call Without Reg Ans Call Without Reg DNS SRY Auto Prefix User ID Use Auth ID Block CID Serv Dist Ring Serv Cfwd Busy Serv Cfwd Sel Serv Block Last Serv DND Serv CWCID Serv Call Back Serv Three Way Conf Serv Unattn Transfer Serv APPENDIX E Configuring Windows Messenger for Use as a SIP User Agent Appendices 1206001236 no v
69. able way for such a UA to discover this mapping However as was noted above the packets may not have an altered post translation port in all cases If the ports are equal a multimedia session will be established without difficulty Unfortunately there are no formal 2000 2006 PortaOne Inc All rights reserved www portaone com 49 Porta SIP System Concepts rules that can be applied to ensure correct operation but there are some factors which influence mapping The following are the major factors e How the NAT server is implemented internally Most NAT servers try to preserve the original source port when forwarding if possible This is not strictly required however and therefore some of them will just use a random soutce port for outgoing connections e Whether or not another session has already been established through the NAT from a different host on the LAN with the same source port In this case the NAT server is likely to allocate a random port for sending out packets to the WAN Please note that the term already established is somewhat vague in this context The NAT server has no way to tell when a UDP session is finished so generally it uses an inactivity timer removing the mapping when that timer expires Again the actual length of the timeout period is implementation specific and may vary from vendor to vendor or even from one version by the same vendor to another NAT and SIP There are two part
70. agement section of the Admin Index page choose IP Phone Profiles 2 Inthe IP Phone Profile management window click the Add icon 3 Fill in the Add IP Phone profile e Name short descriptive name for this profile e Managed By If you plan to use this profile for a certain reseller s customers choose the reseller from the select menu otherwise leave this as Administrator Only e Type The hardware type of the IP phone e As Copy Of This will allow you to create new profiles based on already existing ones for now leave this as None e Effective From Leave the value in this field as immediately 4 Click lel Save IP phone profile settings After clicking lel Save on the previous page you will go to the Profile Edit page where you can edit the generic device settings These configuration parameters are dependent on the specific model of your IP device The example below uses a Sipura 2000 device but most of the settings should be the same for other Sipura Linksys VoIP products 2000 2006 PortaOne Inc All rights reserved www portaone com g 5 Porta SIP Gi H IP Phone Profiles Setting up SIP Services gt Tea seve a savez close cose ot Name Sipura standard i Effective From Type Sipura 2000 Date immediately YYYY MM DD Description Standard profile for residential SIP Time R24 MI 55 Managed By Administrator only Discontinued Enable Web Server Enable W
71. all will be handled by the PortaSIP server in Singapore and the voice traffic will travel only via the Singapore backbone 2 D ITSP PortaBilling PortaSIP Master Slave 2000 2006 PortaOne Inc All rights reserved www portaone com 93 Porta SIP System Concepts This allows VoIP services to be efficiently provided in a situation which is highly typical for many countries or regions good fast Internet connectivity inside the country region and mediocre connectivity with the rest of the world For all users inside that region VoIP traffic signaling and RTP will travel on the local backbone while only small RADIUS packets will travel to the central PortaSwitch location Call flow scenarios for a PortaSiP cluster SIP UA lt gt SIP UA Case A Both SIP phones are registered to the same PortaSIP server SIP Registrations a Porta K Billing Billing Engine PortaSIP PortaSIP In this case the call flow is exactly the same as in a situation where only one PortaSIP server is available discussed earlier in the SIP UA lt gt SIP UA chapter e PortaSIP receives an incoming call and requests authorization and routing from PortaBilline100 e PortaBilling verifies whether this call should be allowed and if the destination is one of our SIP accounts e PortaBilling checks the registration database and returns the address of the PortaSIP server the account is currently registered
72. am for assistance with this configuration task since if you configure the network interface on the SIP server improperly it will render all of your SIP services useless Porta 4 Billing Porta X4 SIP Customers of envA lt lt TT PortaSIP instance j ip smartcall com Environment A m sip smartcall co 195 70 140 2 Environment B z PortaSIP instance sip supercall net 195 70 140 3 Customers of env B Every virtual SIP server acts as an independent PortaSIP installation The virtual SIP instance resides in the var sipenv lt IP gt directory where lt IP gt is the IP address allocated to this SIP instance e g for a PortaSIP working on IP address 120 34 56 78 it will be var sipenv 120 34 256s 78 Inside the sipenv directory there are several sub directories the most important ones being e etc this subdirectory contains a master configuration file for the SIP instance and config files for the individual modules e log PortaSIP log file sip log and copies of the log file for previous days are located here 2000 2006 PortaOne Inc All rights reserved www portaone com 21 Porta SIP System Concepts Clustering of PortaSIP servers You may also install several physically independent PortaSIP servers and connect all of them to the same virtual environment in PortaBilling100 In this case several PortaSIP servers combined in this case into a PortaSIP cluster
73. ame someone example corn Password essees 2000 2006 PortaOne Inc All rights reserved www portaone com 1 29 Porta SIP Appendices 5 To make a call click the Action item in the main menu then select Start Voice Conversation Click the Other tab making sure that Communications Service is selected in the drop down Service box and enter the phone number in the Enter e mail address field as shown below Finally click OR to place a call y Start a Voice Conversation My Contacts Enter the e mail address of the person you want to contact Type the person s complete e mail address f p04 521 527 r Select the service that this person uses SIF Communications Service mes APPENDIX F Setting up a Back to Back T1 E1 Connection Hardware Setup In order to make one or more back to back connections you will need to construct one or more RJ 48C cross over cables using the following table T1 E1 CSU DSU Cross Over Pinout From RJ 48C Pin To RJ 48C Pin ee 2s 5 RE es 2 Make sure you count the RJ 48C pins as shown in the illustration below 2000 2006 PortaOne Inc All rights reserved www portaone com 1 30 Porta SIP Hook underneath S k k y y Appendices Hook underneath h e e i a Eo ES ah Sa Sg SS PRI T1 E1 CrossOver Loopback Cable Altern
74. anagement E Add amp Close bE Logout B Log gt 0 add close fi togout B tog Type Representative Search Direct Customers ANY 4 CDRs Hame Accounts Subcustomers Currency Type Credit limit Balance E mail Status Delete O EasyCall Ltd USD Retail 0 00000 admin easycall com x 2000 2006 PortaOne Inc All rights reserved www portaone com 67 Porta SIP Setting up SIP Services opz Accounts of Retail Customer EasyCall Ltd gt E add E accoun Generator B cose topo Account ID SIP Status v Show Accounts E 4 Add Account for Retail Customer EasyCall Ltd Pe ed save id Save ee coe i tooo Account ID 12061234567 5 Product usp SIP Subscribers Y Blocked gog Opening Balance 10 f Account Info Subscriber Additional Info Life Cycle User Interface Type Debit O Credit O Voucher VolP Password sn9landm E mail Batch easycall New batch Y 2 Next to the customer name click on the amp icon the one in the Accounts column to go to the account management for that customer 3 Click on Add 4 Fill in the Add account form o Account ID SIP ID i e the phone number which will be used to login to the SIP server and receive incoming calls o Product choose the product which you would like your account to have o Blocked you may create your account as blocked alth
75. atively you can order ready made ones You can find a number of vendors producing such cables by searching for RJ 48C cross over cable on www google com Once the cable is ready plug it into the designated pair of T1 E1 ports in your Cisco AS5300 gateway 2000 2006 PortaOne Inc All rights reserved www portaone com 131 Porta SIP Appendices Software Configuration You also have to configure the T1 E1 interfaces The sample configuration below is for T1 adjust the time slots for El isdn switch type primary 5ess controller TI 0 framing sf clock source line primary linecode ami pri group timeslots 1 24 controller TI 1 framing sf clock source line secondary 1 linecode ami pri group timeslots 1 24 controller TL 2 framing sf linecode ami pri group timeslots 1 24 controller Tl 3 framing sf linecode ami pri group timeslots 1 24 interface Serial0 23 no ip address isdn switch type primary 5ess isdn protocol emulate network no cdp enable interface Seriall 23 no ip address isdn switch type primary 5ess no cdp enable interface Serial2 23 no ip address isdn switch type primary 5ess isdn protocol emulate network no cdp enable interface Serial3 23 no ip address isdn switch type primary 5ess no cdp enable 2000 2006 PortaOne Inc All rights reserved www portaone com 1 39 Porta SIP Appendices APPENDIX G SIP devices with auto provisioning available Currently PortaSwitc
76. b 80c3c2c 192 168 1 180 l1efdc57b 80c3c2c 192 168 1 180 Acct Session Time 17 Acct Delay Time Q h323 remote address 66 96 26 134 h323 session protocol sipv2 Acct Terminate Cause User Request 2000 2006 PortaOne Inc All rights reserved www portaone com 99 Porta SIP Setting up SIP Services NAS Port Id 5060 Exec Program Log porta billing pl Jul 21 22 38 24 This call belongs to the environment 1 pb Jul 21 22 38 24 h323 conf id 9C25D01D 9ED3C282 83BAC556 8CE06004 1 call id lefdc57b 80c3c2c 192 168 1 180 1 Jul 21 22 38 24 Found a call in cache with such id Jul 21 22 38 24 Copied account 18667478647758 147059 credit balance 0 55021 limit none of customer PortaOne 2 balance 9996029 96819 limit 0 00000 from 70 68 0 213 into the current request Jul 21 22 38 24 PrepareNexecute GetActiveLegIdByAcct Jul 21 22 38 24 PrepareNexecute UpdateActiveLeg Jul 21 22 38 24 End of the outgoing call for logged in account Waiting another outgoing call or hang up Jul 21 22 38 24 Set lifetime with 15s to Thu dul 21 22 38 39 2005 Jul 21 22 38 24 Looking up vendor connection Jul 21 22 38 24 Trying to match connection for call Jul 21 22 38 24 Looking for a connection VoIP originate Jul 21 22 38 24 Outgoing VoIP matching by the remote IP address 666 96 26 134 env 1 Jul 21 22 38 24 Found connection 306 Free VoIP Network FVN to vendor
77. bK4e23 398ad42b5b3be3602617bd8b10be8c17 0 70 68 0 213 5061 branch z9hG4bK4de91149a7529218ce70fb9209be2ef9 rport 5 Via SIP 2 0 UDP 061 Max Forwards 16 From lt sip 380443333333 70 68 0 213 gt tag 54cce4d15b86cbhbe9dec50522083bflc TO Call ID CSeq 100 BYE Contact Expires 300 User Agent cisco GUID h323 conf id Sippy 758 lt sip 18667478647758 70 68 0 213 gt tag a4044d2fe886380800 lefdc57b 80c3c2c 192 168 1 180 Anonymous lt sip 70 68 0 213 5061 gt 2619723805 2664678018 2210055510 2363514890 2619723805 2664678018 2210055510 2363514890 B2BUA sends stop accounting to the billing for the incoming call leg 21 Jul 23 38 24 1lefdc57b 80c3c2c 192 168 1 180 b2bua Answer User Name Calling Station Id Called Station I Id h323 call origin h323 call type h323 conf idqd call id Acct Session Id h323 remote address h323 session protocol h323 setup time h323 voice quality Acct Terminate Cause h323 ivr out h323 disconnect time h323 connect time Acct Session Time h323 disconnect cause Acct Status Type sending Acct Stop 18667478647758 18667478647758 380443333333 answer VOLE 9C25D01D 9ED3C282 83BAC556 8CEOQ600A lefdc57b 80c3c2c 192 168 1 180 lefdc57b 80c3c2c 192 168 1 180 None S10v27 06 37 26 000 GMT Fri Oo User Request PortaBilling Session 06 38 23 000 GMT Fri 06 38 06 000 GMT Fri Y7 wAY Stop Jul 22 2005
78. be used to prevent hammering the SIP server with registrations every second Of SO REG EXPIRES MAX Maximum time interval during which the registration will be considered valid in seconds defaults to 7200 any non NAT UA The default is 0 RTPP_LOCAL POLICY RTPP local policy for SIP to SIP calls for this node overriding the policy configured in BE the default is unset After you have modified the porta sip conf file for a certain SIP instance you must restart that instance sudo var sipenv lt ip gt etc rc d sip sh restart 2000 2006 PortaOne Inc All rights reserved www portaone com 1 1 9 Porta SIP Administration FAQ PortaSIiP related parameters in porta billing conf Since routing is controlled by PortaBilling100 some of the configuration parameters may be modified in the configuration file for the billing engine home porta billing etc porta billing conf on the PortaBilling100 master server Routing RTPPLoOcal Policy RTPPRemotePolicye RTP proxy policy for SIP to SIP calls RTPPLocalPolicy is a policy enabling use of an RTP proxy for calls between SIP phones The possible options are 1 direct do not use RTP proxy 2 nat use RTP proxy if either of the UAs is behind NAT 3 all always use RTP proxy By default the RTP policy for SIP to SIP calls is set to nat which enables reliable communication between SIP phones while simultaneously avoiding unnecessary RTP proxying R
79. ca Vancouver v Password auto Web Interface Language en English i Access Level Vendor v Output Format Date vvyy MM DC 2003 12 31 Time HH24 MI SS User Defined Date amp Time YYYY MM DD HH24 MI 5S User Defined Input Format Date rYYY MM DD 2003 12 31 Time HH24 MI SS User Defined 3 Fill in the Add Vendor form Please note that there are two tabs available on the screen The most important fields are Main form top o Vendor name short name for the Vendor object this will be used on the web interface o Currency the currency in which this vendor charges you O Opening balance starting balance for the vendor the default is zero Additional info o Billing period split period for vendor statistics User Interface o Time zone the time zone that the vendor uses for his billing period when sending you an invoice Statistics will be split into periods in this time zone so your statistics will match the vendor s 4 Click lel Save 5 If you plan to terminate your calls to the vendor s SIP server typically he would provide you with a username password which will authorize you to send calls to his server Enter this information as Vendor account 2000 2006 PortaOne Inc All rights reserved www portaone com 6 4 Porta SIP Setting up SIP Services E 4 Edit Vendor OEE Gl Save Cos cons 82 Co
80. call quality since the RTP stream follows the standard path SIP phonel gt SIP server gt SIP phone2 2000 2006 PortaOne Inc All rights reserved www portaone com 26 Porta SIP System Concepts SIP UA gt PSTN When a SIP phone user makes a call to an off net destination only one PortaSIP server and PortaBilling are involved in the call flow So this works in exactly the same way as described earlier for SIP gt PSTN calls in the case of a single PortaSIP server Porta 4 Billing Billing Engine Billing Provisioning I I PSTN gt SIP UA Again the call flow is extremely similar to the usual PSTN gt SIP call flow The gateway delivers a call to a PortaSIP server which then sends the call to the SIP phone Porta K4 Billing Billing Engine Billing Provisioning I i 2000 2006 PortaOne Inc All rights reserved www portaone com 26 Porta SIP System Concepts SIP phone configuration for PortaSIP cluster In order to ensure reliable VoIP services a SIP phone must be able to automatically switch to backup servers should one of the SIP servers not be available How does a SIP phone know about alternative SIP servers There are several options 1 Program the backup SIP server s IP address into the SIP phones if this is supported by the IP phone configuration The main disadvantage of this method is that it only works with certain SIP phone models 2 Instead of programming th
81. ccounts and terminate calls from my SIP accounts to PSTN How many simultaneous sessions will it be able to handle tule of thumb is that each SIP gt PSTWN call or PSTN gt SIP call will use up one DSP and one timeslot in E1 T1 interface Therefore if you have connected your gateway to PSTN using for example two E1 ports and are using both of those ports for SIP lt gt PSTN the maximum number of 2000 2006 PortaOne Inc All rights reserved www portaone com 1 1 5 Porta SIP Administration FAQ simultaneous calls you will be able to handle will be 60 provided that you have enough free DSPs in the system have problems with the audio quality of SIP calls what can do First of all please make sure that both the user agents and SIP lt gt PSTN gateway are configured for use of the same low bitrate codec such as G 723 In APPENDIX B Cisco GW Setup for PortaSIP COMEDIA there are details on how to configure Cisco IOS and Cisco ATA 186 for other SIP phones or gateways check the documentation supplied with the device If you are sure that the codec used for SIP calls is a low bitrate one for example by inspecting the gateway logs but the quality is still suboptimal you need to determine where packet loss is occurring in the media path To do this you can use standard network tools such as ping traceroute and the like Keep in mind that for SIP UA lt gt PSTN calls the RTP audio stream flows directly between
82. ce and click Upload on the Destinations screen i 4 Destinations Upload gt a save id Save ee coef i tooo File C Demo DefaultDestinationsSet csy Browse _ 6 Type in the filename for the file you have edited or click on the Browse button and select the file 2000 2006 PortaOne Inc All rights reserved www portaone com 53 Porta SIP Setting up SIP Services 7 Click ml Save amp Close Destinations for SIP phones In order to receive an incoming call an SIP user agent must be configured with a phone number Normally you will obtain a range of phone numbers from your local telecom and you will be able to assign these to your customers For example you will be assigned range 12027810000 12027819999 It is therefore a good idea to create a special destination 1202781 This prefix will cover all of your SIP phones and thus its actual purpose is to set up your pricing or routing Even if you have not obtained an official phone prefix it is highly recommended not to assign IDs to your SIP user agents at random Choose a non existing prefix e g 777 and create it as the destination with N A country and the description SIP phones Then use SIP IDs such as 77700001 7770002 7770999 Create Nodes Now you have to enter your SIP server and optionally other gateways as nodes PortaBilling requires some key information about your network equipment such as the IP address h323 id
83. currencies define these in the Currencies section and specify exchange rates in Exchange Rates E A Currencies E Add M Save fp Save amp Close amp Close PE Logout gt Badd tel Save fel Save amp Close close og Search ISO 4217 Dec 3 Exchange Rate Payment Minimum Edit wane mn Hame digits Major Minor eS System Payment Method P TEN Delete v v E CZK 203 Czech Koruna 2 koruna haler XE com Authorize net Bank account eCheck 100 00000 x Choosing XE COM as exchange rate source you agree to the XE COM Terms of Use Create Destinations This step is only required if you have not previously defined the necessary destinations There are two ways to insert a new destination into the system e One by one using the I Add functionality on the web interface e A bulk update by uploading destinations from a file NOTE PortaBilling supplies a file with a set default destination which you can download and then upload to the server However it is possible that your business requires different types of prefixes so please check the data in the file before uploading Creating destinations one by one 1 Inthe Management section of Admin Index choose Destinations 2 Click on the amp Add button 2000 2006 PortaOne Inc All rights reserved www portaone com 59 Porta SIP Setting up SIP Services 3 Fill in the required information This i
84. d www portaone com Acct Session Time 17 Acct Delay Time 0 h323 remote address None h323 session protocol sipv2 Acct Terminate Cause h323 ivr out NAS Port Id Exec Program Log User Request POortaBilling Sess1on unliock 5060 borta billing pl Jul 21 22 38 24 This call belongs to the environment 1 pb Jul 21 22 38 24 h323 conf id 9C25D01D 9ED3C282 83BAC556 8CE0600A 1 call id lefdc57b 80c3c2c 192 168 1 180 1 Jul 21 22 39 24 Found a call in cache with such id Jul 21 22 38 24 Copied account 18667478647758 147059 credit balance 0 63051 limit none of customer PortaOne 2 balance 9996029 88789 limit 0 00000 from 70 68 0 213 into the current request Jul 21 22 38 24 PrepareNexecute GetActiveLegIdByAcct Jul 21 22 38 24 PrepareNexecute UpdateActivelLeg Jul 21 22 38 24 Force unlock requested by NAS Jul 21 22 38 24 Scheduling 18667478647758 for Logout call lifetime reduced to 15 Jul 21 22 38 24 Logging out account 18667478647758 147059 from 9C25D01D 9ED3C282 83BAC556 8CE06004 Jul 21 22 38 24 Set lifetime with 15s to Thu Jul 21 22 38 39 2005 Jul 21 22 38 24 Looking up vendor connection Jul 21 22 38 24 Trying to match connection for call Jul 21 22 38 24 Looking for a connection VoIP answer Jul 21 22 38 24 No original CLD using CLD from the request Jul 21 22 38 24 VoIP matching by the node IP 70 68 0 213
85. d then be described as a VoIP to vendor connection in the billing with the remote address being the address of the vendor s SIP server or SIP enabled gateway e The billing engine returns the IP address of the vendor s SIP server in the route information with login password optional The PortaSIP server sends an INVITE request to that address providing the proper credentials and then proceeds in basically the same way as if it were communicating directly with C s SIP user agent e After the call is established the B2BUA starts the call timer disconnecting the call once the maximum call duration is exceeded e After the call is completed the BZBUA sends accounting information for the call to the billing 2000 2006 PortaOne Inc All rights reserved www portaone com 1 6 Porta SIP System Concepts Terminating SIP calls to a vendor using telephony Porta 4 Billing SIP phone A GW NY 02 Phone C 12 34 56 78 e Let s assume that T1 is connected to Qwest on our gateway GW NY 02 in New York where we are able to terminate calls to the US This connection would be described as a PSTN to vendor connection The PortaSIP server obtains the address of the GW NY 02 gateway in the route information e The B2BUA sends an INVITE to the remote gateway GW NY 02 e GW NY 02 performs authentication on the incoming call via the remote IP address Even if the call was actually originated by A a dynamic IP address but th
86. ded to You may use a popup window to search for a specific account Also note that To Dial may contain any phone number e g your partner s mobile phone number and not just one of the SIP account IDs NOTE If you enter an off net PSTN number in To Dial it must be in the E 164 format i e you cannot enter the number in the customer s dialing format 4 Click the l Save button in the toolbar or the ml icon on the left side of the row 5 Repeat steps 2 4 to add all the required abbreviated numbers E 4 Edit Customer fe lsat ER save ER Savet tiore coe BR cors Account tot Eu Customer Name EasyCall Ltd Opening Balance 0 00000 USD L Blocked go Balance 0 00000 USD E Type Retail a Li L a a dit Abbreviated To Dial Description SIP Delete E po W e E E 101 12961234567 Mary Smith Q x El Select Account e ease O Account ID Batch Ctrl OOOO omen Account ID Batch Status SIP 12061234567 easycall 12061234568 easycall 12061234569 easycall 12061234570 easycall 2000 2006 PortaOne Inc All rights reserved www portaone com 71 Porta SIP Setting up SIP Services Configure Cisco ATA Using ATA Expert optional Cisco ATA could be configured from the web interface accessible at http lt ata IP address gt dev However this web interface is designed to be used by experts and parameter values must be entered in the protoc
87. e INVITE request to GW NY 02 arrived from the PortaSIP server the PortaSIP s IP address will be authenticated Since PortaSIP is defined as our node authentication will be successful NOTE Remote IP authentication on the gateway is not required in this case but is highly recommended Otherwise someone else might try to send calls directly to the gateway bypassing authentication and making such calls for free e The call will be routed to the PSTN on the gateway e After the call is established the B2BUA starts the call timer disconnecting the call once the maximum call duration is exceeded e After the call is completed the B2BUA sends accounting information for the two VoIP call legs to the billing The gateway will also send accounting information about the answer VoIP and originate Telephony call legs The billing engine will combine this information since accounting from the SIP server allows us to identify who made the call while accounting from the gateway carries other useful information for example through which telephony port the call was terminated 2000 2006 PortaOne Inc All rights reserved www portaone com 1 7 Porta SIP System Concepts PSTN gt SIP Phone C GW NY 01 Q esm 2 Porta K Billing 5 6 SIP phone A This is another important aspect of SIP telephony Your subscribers not only want to make outgoing calls they also want other people to be able to call them on
88. e IP address of the SIP server into the SIP phone s config use a hostname instead e g sip supercall com This name can be set up to resolve to multiple IP addresses of different SIP servers DNS round robin However this may not work if the manufacturer of the SIP phone has employed a simplified approach so that the phone does not perform DNS resolving if a current SIP server fails 3 Use the DNS SRV records These records were designed specifically for the purpose of providing clients with information about available servers including the preferred order in which individual servers should be used in a redundant multi server environment This method is currently the most flexible and reliable one see details below Using DNS SRV records for multiple PortaSIP proxies an example Here we assume that you have two PortaSIP servers available in the main hosting center for your VoIP mysipcall com service as well as one backup PortaSIP server in a collocation center in a different city Your users normally use either one of the main servers and only if they cannot access either of them e g a network problem affecting the entire hosting center will they go to a backup one First of all your DNS server for the mysipcall com domain must be configured with DNS A records for the individual PortaSIP servers portasipl IN A eS Re ral 610 res res portasip2 IN A 199 100 5 portasip3 IN A 64 17 6S 227 After this
89. e event 8000 a fmtp 101 0 15 Start accounting for the incoming answer VolP call leg is sent to the billing 21 Jul 23 38 07 1efdc57b 80c3c2c 192 168 1 180 b2bua sending Acct Start Answer User Name 18667478647758 Galling Stalion id 1866 7478647758 Cal Led sStation id 380443333333 h323 call origin answer h323 call type VoIP h323 conf id 9C25D01D 9ED3C282 83BAC556 8CEO600A call id lefdc57b 80c3c2c 192 168 1 180 Acct Session Id Jefdc57b 80c3c2cQ 192 168 1 180 h323 remote address None h323 session protocol sipv2 h323 setup time 06237226 000 GMT Fri Jul 22 20057 h323 connect time 06 38 07 000 GMT Fri Jul 22 2005 Acct Status Type Start The caller s SIP UA acknowledges the call connect 2000 2006 PortaOne Inc All rights reserved www portaone com 83 Porta SIP Setting up SIP Services 21 Jul 23 38 07 GLOBAL ser RECEIVED message from 62 244 32 30 50563 ACK sip 70 68 0 213 5061 SIP 2 0 Via SIP 2 0 UDP 192 168 1 180 5060 branch z 9hG4bK 9d34a341 From 758 lt sip 18667478647758 70 68 0 213 gt tag a4044d2fe886380800 To lt sip 380443333333 70 68 0 213 gt tag 54cce4d15b8 6cbhbe9dec50522083bflc Call ID l1efdc57b 80c3c2c 192 168 1 180 CSeq 102 ACK Max Forwards 70 Route lt sip 70 68 0 213 ftag a4044d2fe886380800 1r gt Authorization Digest username 18667478647758 realm 70 68 0 213 nonce 5dbe8444a6063753 3 5c247b
90. eb Admin Access amnes User Password O O Field Description Enable Web Enable disable the built in web server for device Server monitoring Enable Web Enable disable the built in web server for device Admin Access administration and changing configuration parameters Admin Password The password for administrator access User Password The password for user access m IP Phone Profiles Ce E save savexciose aise OOOO ooe Name Sipura standard 7 Ti Effective From Type Sipura 2000 Date immediately _ VVVV MM DD Description Standard profile for residential SIP Time HN 4 M55 Managed By Administrator only Discontinued Profile Rule key B htto PB SLAVE SERVER EA EMA cfg GPP A OPP B ge Profile Rule Path to the profile configuration file Replace PB SLAVE SERVER with the IP address or hostname of your provisioning server PortaBilling slave server The example on the screen assumes provisioning via the HTTP protocol change http to tftp for TFTP provisioning MA will be replaced by the concatenated MAC address from the IP Phone Inventory so leave it as is 2000 2006 PortaOne Inc All rights reserved www portaone com 56 Porta SIP Setting up SIP Services GPP A GPP B Dynamic variables During config file generation GPP C i_env will be replaced by a unique environment ID and ascii_key by the crypt key from the IP phone There is no need to change any
91. el Save el Save amp Close amp Close PT Logout amp Log gt i Save im Save amp Close Close Ss Logout B Log Name Sipura Description Test sipura Managed by Administrator only Type Sipura 2000 TIR General Info Profile Sipura standard M MAC Address 00 06 08 AB D7 Aq Ports Total 2 Free ASCII Key v 2000 2006 PortaOne Inc All rights reserved www portaone com 98 Porta SIP Setting up SIP Services Gi IP Phone Inventory gt E ad cose dog B Lo Search axr z Caa Hame Type Managed By MAC Address Description Delete Sipura Sipura 2000 Administrator only 0 E 8 AB D7 A8 Test sipura x 1 In the Management section of the Admin Index page choose IP Phone Inventory 2 Inthe IP Phone Inventory management window click the Add icon 3 Fill in the Add IP Phone form e Name A unique ID for the IP phone e Type The hardware model of the phone e Profile The IP phone profile you previously created e MAC Address The hardware ID of the IP phone typically printed on the back of the device 6 hexadecimal numbers separated by colons Make sure you enter the full value of the MAC address including the colons e Ports How many phone lines are available on this device 4 Click ml Save amp Close Provisioning an account on an IP phone
92. en Premium routing categories e Cost efficient includes only Normal routing category 2000 2006 PortaOne Inc All rights reserved www portaone com 3 4 Porta SIP System Concepts So depending on which routing plan is assigned to the current customer the system will offer a different set of routes Routing algorithm The routing principle is simple e The SIP server or MVTS or some other entity asks PortaBilling for routing destinations for a given number e PortaBilling checks every tariff with routing extensions associated with a vendor connection for rates matching this phone number In each tariff the best matching rate is chosen this rate will define the routing parameters e A list of possible termination addresses will be produced this will include the remote IP addresses for VoIP connections and IP addresses of your own nodes with telephony connections e This list will be sorted according to routing plan routing preference and cost entries after the first huntstop will be ignored e A list of these IP addresses with optional login and password for SIP authentication will be returned to the SIP server To avoid extremely long delays only a certain number of routes from the beginning of the list are returned the default is 15 but this can be changed in porta billing conf Route sorting How exactly does PortaBilling100 arrange multiple available routes 1 By route category Only route categories w
93. enerated characters The SIP UA must compute a response using this challenge a username a password and some other attributes with the MD5 algorithm This response is then sent back to the SIP server in another INVITE request The main advantage of this method is that the actual password is never transferred over the Internet and there is no chance of recovering the password by monitoring challenge response pairs Such digest authentication provides a secure and flexible way to identify whether a remote SIP device is indeed a legitimate customer Authorization based on IP address Unfortunately some SIP UAs e g the Cisco AS5300 5350 gateway do not support digest authentication for outgoing calls This means that when the SIP UA receives a 401 Unauthorized reply from the SIP server it will simply drop the call as it is unable to proceed with call setup In this case PortaSIP may be configured to detect that a call is coming from a digest incapable SIP UA and perform authorization based on the SIP UA s remote IP The User Name attribute in the RADIUS authorization request will contain the remote IP address and if an account with such an account ID exists in the billing database and this account is allowed to call the dialed destination the call will be permitted to go through ip auth table in porta sip database describes various ways to detect such SIP UAs It contains different patterns which may be applied to various pa
94. ently registered SIP user agents Thus it is able to determine how to contact a specific SIP user agent if there is an incoming call In response to the authorization request the billing engine informs the SIP server that the dialed number is actually a valid SIP account and that the call should be sent to the SIP user agent Note that routing the call to a SIP user agent is only one of the possible routes for instance a call can be redirected to follow me numbers or a unified messaging service if the account is not available online at the moment Routing of SIP Off net Calls You can have different vendors for terminating off net calls For example calls to the US can be terminated either to AT amp T via a T1 connected to yout gateway in New York or by sending the call to a remote gateway from Qwest You need a tool allowing you to manage routing policies for the different destinations This tool is extensions routing for tariffs Tariffs define the termination costs for each connection to a vendor while extensions routing simply adds a few more fields to the rates in a given tariff This allows you to easily manage both termination costs and routing from a single location on the web interface The routing principle is simple e The SIP server asks PortaBilling for routing destinations for a number e PortaBilling checks every tariff with routing extensions associated with connection to the vendor for rates matching this phone n
95. ers assigned to FXS ports or IP phones Therefore if you have a gateway with E1 T1 connected to it and wish to route certain prefixes there for termination you must define the routing in the billing To do this proceed as follows e Create a new tariff with the Routing Ext 2000 2006 PortaOne Inc All rights reserved www portaone com 1 07 Porta SIP How to e When you enter rates into this tariff two new columns will appear Preference and Huntstop Enter the desired routing preference The higher the number the more desirable this route is 0 means no route at all Turn the huntstop on if you do not wish to use any routes with a lower priority e Create a PSTN to vendor connection to the vendor specify the gateway which will handle termination as your Node and select the tariff you have created as the termination tariff e Make sure that your gateway is actually configured to accept incoming VoIP calls and send them to telephony for the destinations you plan to terminate allow my customer to have two phone numbers from different countries which will both ring on the same SIP phone You can have an unlimited number of such extra phone numbers Your customer will have one main account e g 12027810003 which will be provisioned on his phone plus some extra accounts e g 4981234567 with the follow me service on these accounts configured to always go to 12027810003 create an applica
96. erved www portaone com 5 Porta SIP PortaSIP User Guide Installation In order to simplify installation and support as much as possible PortaSIP is provided on a jump start installation CD This CD contains installation media for FreeBSD 5 4 stable branch with the latest security bug fixes supplementary packages necessary for convenient system administration and maintenance and PortaSIP software packages PortaSIP installation and configuration are automated and integrated within the main installation process This allows you to install a completely functional PortaSIP server from scratch in less than 15 minutes For detailed installation instructions please refer to the PortaSIP Installation Guide What s New in Maintenance Release 11 This release includes several new features and improvements e Automatic welcome message the customer s phone will ring after he connects to your service for the first time and he will hear your custom greeting e Clustering of PortaSIP servers you can install several PortaSIP servers in different locations for better network utilization and redundancy SIP phones can use any of the available PortaSIP servers e Inventory of IP devices IP phones and adaptors new supported types of devices for auto provisioning e New IP PBX feature customizable music on hold e Ability to browse SIP log files from the PortaBilling web interface e Multi lingual announcements from the media
97. es offline during a phone conversation e g an Internet line is down the SIP server may not be aware of this fact Thus if the remote party does not hang up e g there is an automated IVR or a problem with disconnect supervision this call may stay in the active state for a long time To prevent this situation PortaSIP has a keep alive functionality available e Customer A tries to call B and the call is connected e While the call is in progress PortaSIP periodically sends a small SIP request to the SIP phone e Ifthe phone replies this means that the phone is still online e If no reply is received PortaSIP will attempt to resend the keep alive packet several times this is done to prevent call disconnection in the case of an only temporary network connectivity problem on the SIP phone side e If no reply has been received following all attempts PortaSIP will conclude that the SIP phone has unexpectedly gone offline and will disconnect the other call leg and send an accounting record to the billing e Therefore the call will be charged for a call duration quite close to the real one First login greeting This is a feature not directly related to call processing but is something that will give your PortaSwitch based VoIP service a competitive advantage When a customer unpacks his new SIP phone and connects it to the Internet the phone will start ringing When the customer picks up the phone he will hear a greet
98. eserved www portaone com 55 Porta SIP Mes ww Setting up SIP Services Add Tariff gt E save tel savenciore coe Logo Name SIP Phone subscribers Currency USD US Dollar v Type Ordinary Y General Info DE Off Peak Period startstop hr 20 5 Off Peak Description PERIOD From 20 00 until 06 00 Destination Group Set Free Seconds Post Call Surcharge Login Fee Connect Fee Round Charged Amount xxxxx xxO000 Formula Short Description SIP phone subscriber s tariff Description This tariff applies to all SIP phone subscribers 1 In the Management section of the Admin Index page choose Tariffs 2 On the Tariff Management page choose H Add 3 Fillin the Add Tariff form O O Name a short name for the tariff object this is the name you will then see in the select menus Currency Indicates in which currency pricing information is defined All pricing information for a single tariff must be defined in the same currency NOTE The currency for the tariff may be chosen only once and cannot be changed later Type If this is a tariff that describes your vendor s termination costs choose Routing here as this tariff will be used not only to calculate termination costs but also for routing SIP calls Off peak Period Defines the off peak period Click on the Off peak period wizard icon CA to su
99. et key in the GPP B field 5 Apply the changes by clicking Submit All Changes at the bottom of the page After reboot the device will request the configuration file from the server via the http protocol and all profile parameters will be applied Advanced Provisioning Tips In order to understand provisioning in depth you have to know something about the internal processes of configuration file generation Auto provisioning is a built in function of an IP phone allowing the device to download its configuration from an external server via the HTTP or TFTP protocols Different IP phones use a different set of configuration parameters and a different format for the configuration file 2000 2006 PortaOne Inc All rights reserved www portaone com 1 01 Porta SIP Setting up SIP Services In addition many IP phone manufacturers require the configuration file to be processed by some proprietary utility profile compiler before it can be supplied to the IP phone PortaBilling initially processes each account with an associated IP phone and creates a parameter value plain text file located in usr home porta admin profile This file then becomes the source for the manufacturet s configuration compiler The result of the compilation i e the file to be downloaded by the IP phone is placed in usr home porta admin apache For convenience in the initial setup usr home porta admin apache htdocs is the root directory for the defa
100. etween the public WAN address the packet was sent to and the private LAN address it was received from so that when the reply comes it can carry out a reverse translation i e replace the public destination address of the packet with the private one and forward it to the destination on the LAN Since the NAT server can potentially map multiple private addresses into a single public one it is possible that a TCP or UDP packet originally sent from for example port A of the host on the private LAN will then after being processed in the translation be sent from a completely different port B of the NAT s WAN interface The following figure illustrates this here HOST 1 is a host on a private network with private IP address 1 2 3 4 NAT is the NAT server connected to the WAN via an interface with public IP address 9 8 7 6 and HOST 2 is the host on the WAN with which HOST 1 communicates IP 1 2 3 4 IP 9 8 7 6 Port 56789 Port 12345 problem relating to the SIP User Agent UA arises when the UA is situated behind a NAT server When establishing a multimedia session the NAT server sends UDP information indicating which port it should use to send a media stream to the remote UA Since there is a NAT server between them the actual UDP port to which the remote UA should send its RTP stream may differ from the port reported by the UA on a private LAN 12345 vs 56789 in the figure above and there is no reli
101. example if you plan to charge your customers more when they access a toll free line instead of a local one you will need two tariffs i e Normal and Using Toll free line These tariffs should be created using the Ordinary type e Create a tariff with the termination costs for each termination partner you have these tariffs will also include your routing preferences e Ifyou have resellers create the tariffs you will use to charge each of them Do not create tariffs which will be applied to subscribers of your resellers yet Create customers first and then return to this step Make sure that when creating these subscriber tariffs you choose the Managed by NNN in the Type menu where NNN is the name of the cotresponding resellers 2000 2006 PortaOne Inc All rights reserved www portaone com 61 Porta SIP Setting up SIP Services Create Product Accounts for accessing your SIP services will be issued for a specific product Products are a powerful feature that defines different ways to bill an account Product definition is always done in two steps product definition and creation of an accessibility list i 4 Product Management gt Badd cose o y Oj Bo Managed By Search ES Jj search Name Currency Managed By Description Delete E 4 Add Product gt Tid save cose Tm ogo Product Name sp subscribers S Currency usp US Dollar x Managed By Administra
102. g SIP phone A SIP phone B Offline or Not Answering User dials 121 in an attempt to reach user B His SIP user agent sends an INVITE request to the SIP server 1 The SIP server performs authorization in the billing 2 The billing will perform number translation and determine whether the destination number is actually an account The billing checks the registration database but finds that this account is not online at the moment If B has unified messaging services enabled the billing will return routing 3 for this call which will be sent to the UM gateway Thus A will be redirected to a voicemail system and can leave a message for B 6 The same thing would happen if B were online but not answering his phone 4 5 In any other case the call will fail 2000 2006 PortaOne Inc All rights reserved www portaone com 1 3 Porta SIP System Concepts Call between several PortaSIP servers You can use several PortaSIP servers simultaneously for improved reliability or better network utilization Let s assume you have two PortaSIP servers the primary one in New York and a second one installed in Frankfurt The Frankfurt PortaSIP serves most of your European customers i e they connect to it via the fast intra European IP backbone and acts as a backup for all other users around the world Thus the SIP phone will try to register there if the New York 1 s server is down or for some reason inaccessible Po
103. g on the situation For instance when your SIP account 12027810003 makes an off net call to the United States PSTN network the ANI number must be in the 10 digit format area code phone number 1 e 2027810003 This is accomplished via the CLI translation rule property of the vendor s connection CLI translation rules calls terminated to SIP phones Another extremely useful feature of the CLI translation rule is PortaSwitch s ability to convert the CLI ANI number for the incoming call into the customer s dialing format activated in the customer s dialing rules settings Let s assume that a customer has a SIP phone with the phone number 12027810003 provisioned to it and his dialing rules are setup for North America While out for lunch he receives three calls e From phone number 12027810002 his colleague e From 420298765432 his customer in the Czech Republic e From 12061234567 his old friend from Seattle The ANI CLI numbers for all these calls will be converted so that when he returns from lunch he will see e 7810002 e 011420298765432 e 12061234567 Now he can simply hit redial on his phone to initiate a call since these numbers are already in the same format as he would have normally dialed 2000 2006 PortaOne Inc All rights reserved www portaone com 40 Porta SIP System Concepts Routing of SIP On net Calls The SIP server automatically maintains information about all curr
104. guring Windows Messenger for Use as a SIP User PR OMG E EE av veneer E E A AE OAN acinus A I 127 APPENDIX F Setting up a Back to Back T1 E1 Connection 130 APPENDIX G SIP devices with auto provisioning available uu 133 2000 2006 PortaOne Inc All rights reserved www portaone com 3 Porta SIP PortaSIP User Guide Preface This document provides PortaSIP PortaSwitch users with the most common examples and guidelines for setting up a VoIP network The last section of the document answers the most frequent questions users ask after running PortaSwitch for the first time Where to get the latest version of this guide The hard copy of this guide is updated at major releases only and does not always contain the latest material on enhancements occurring in between minor releases The online copy of this guide is always up to date integrating the latest changes to the product You can access the latest copy of this guide at www portaone com resources documentation Conventions This publication uses the following conventions Commands and keywords are given in boldface Terminal sessions console screens or system file names are displayed in fixed width font Caution indicates that the described action might result in program malfunction or data loss NOTE Notes contain helpful suggestions about or references to materials not contained in this manual Timesaver means that you can save time by performing the actio
105. h can auto provision the following SIP phones ATAs e Cisco ATA 186 firmware versions 2 and 3 e Sipura 1001 e Sipura 2000 e Sipura 3000 e Linksys PAP2 e Linksys WRT54GP2 e GrandStream HT486 e GrandStream HT496 2000 2006 PortaOne Inc All rights reserved www portaone com 1 33
106. heck the box if this account has unified messaging e g voicemail services enabled o Follow Me Enabled check the box if this account has follow me feature enabled If yes account owner can define a list of the numbers where the incoming call to his UA will be redirected for example his home phone mobile Life Cycle tab O Activation date account activation date Expiration date account expiration date o Life time Relative expiration date account will expire on first usage date life time days If you do not want to use this feature leave the field blank O User Interface tab Oo Login Account login to web self care pages Can be the same as account ID Oo Password password for the web self care pages o Time zone When an account owner pre paid card user accesses web self care pages to see a list of his calls we can show the time in the time zone most appropriate for him 5 After clicking l Save amp Close you will see a confirmation screen saying that the new account has been created 2000 2006 PortaOne Inc All rights reserved www portaone com Porta SIP Setting up SIP Services Set up Dialing Rules for the Customer optional It could be that your customer wishes to use his custom numbering format For example in order to make transition from PSTN PBX to VoIP as easy as possible he requires that his users should be able to dial the phone number
107. hich are included in the routing plan will be used following the order given in the routing plan 2 Ifyou have multiple route categories within the routing plan you can either merge them into the same group by assigning them the same order value or keep each one separate with its own order value Then routes within the same order group for route categories will be arranged according to preference 3 For routes with the same preference the system can arrange them according to cost a comparison is made on the Price_Next rate parameter so that cheaper routes will be among the first ones or in random fashion Does PortaSwitch support LCR Yes we support LCR and much mote besides In fact just LCR is the simplest type of routing PortaSwitch handles If you decide not to use routing plans one default plan for everyone or routing preferences same preference for all vendors then the routing decision will be affected solely by the vendor s cost 2000 2006 PortaOne Inc All rights reserved www portaone com 35 Porta SIP System Concepts Compare the two illustrations below The first one shows simple least cost routing all of the available carriers are arranged according to cost In the second illustration PortaSwitch routing preferences are used so that the administrator can re arrange the routing In this case carrier D has been moved down the routing list while carrier A has been moved up to the to
108. i Jul 22 2005 Acct Status Type Start Media traffic from the called party is directed to the RTP proxy and information about call connect is sent to the caller 21 Jul 23 38 07 GLOBAL rtpproxy received command L lefdc57b 80c3c2c 192 168 1 180 66 96 26 134 20620 813da13c69239f8d427bbd3b78c601ee 5q911c473909f6qd59b54af2f14f12986 21 Jul 23 38 07 1efdc57b 80c3c2c 192 168 1 180 rtpproxy lookup on a ports 35134 35136 session timer restarted 21 Jul 23 38 07 1lefdc57b 80c3c2c 192 168 1 180 rtpproxy pre filling callee s address with 66 96 26 134 20620 21 Jul 23 38 07 GLOBAL rtpproxy sending reply 35136 70 68 0 213 21 Jul 23 38 07 GLOBAL ser SENDING message to 62 244 32 30 50563 SIP 2 0 200 OK Viar SIP 2 0 UDP 192 168 1 180 5060 received 62 244 32 30 rport 50563 branch z9hG4bK A43b706 Record Route lt sip 7 0 68 0 213 ftag a4044d2fe886380800 1r gt From 758 lt sip 18667478647758 70 68 0 213 gt tag a4044d2fe886380800 To lt sip 380443333333 70 68 0 213 gt tag 54cce4d15b8 6cbhbe9dec50522083bflc Call ID l1efdc57b 80c3c2c 192 168 1 180 CSeq 102 INVITE Server Sippy Contact Anonymous lt sip 70 68 0 213 5061 gt Content Length 263 Content Type application sdp v 0 o 1122014231 1122014231 IN IP4 66 996 26 134 es t 0 0 m audio 35136 RTP AVP 18 0 4 8 101 c IN IP4 70 68 0 213 a rtpmap 18 G729 8000 a rtpmap 0 PCMU 8000 a rtpmap 4 G723 8000 a rtpmap 8 PCMA 8000 a rtpmap 101 telephon
109. ic codec conversion there are three files for each prompt pcm 723 and 729 These files are located in usr local share asterisk sounds and you can change them according to your needs Here is a list of the currently supported error types e account_expired the account is no longer active expired as per the expiration date or life time e cld_blocked there was an attempt to call a destination which is not in the tariff or is marked as forbidden e credit disconnect a call is disconnected because the maximum credit time is over e in_use this call attempt is blocked because another call from the same debit account is in progress e insufficient_balance there are not enough funds to make a call to the given destination e invalid_account incorrect account ID or account is not permitted to use SIP services e user denied the account is blocked e wrong passwd an incorrect password has been provided Every account in PortaBilling has a preferred language property which defines the desired language for VRs The language code e g ch for 2000 2006 PortaOne Inc All rights reserved www portaone com 30 Porta SIP System Concepts Chinese assigned to the account is returned from the billing so the media server will first attempt to play a voice prompt for that language If that prompt does not exist the default English voice prompt will be played Keep alive call monitoring If a SIP phone go
110. ijing 0 09000 USD Default 64 67 2191 VendorF Termination to vendor F Vendor F 5 86 CHINA Proper 0 11000 USD Default 193 50 123 6 VendorE Termination to carrier E Vendor E 6 86 CHINA Proper 0 02500 USD Default 45 12 156 200 Yendor D Termination to vendor D Vendor D Routing configuration example Tariff A 8610 0 04 min Cheap 7 Tariff B Tariff C 86 0 06 min 86 0 03 min Default 5 N Termination Partner C Termination Partner A N Termination Partner B Termination Termination Partner D F Partner F nin 86 0 025 n ia oO Consider the following example If you have 1 A Standard routing plan which includes three route categories Default order 70 Cheap order 40 and Expensive order 10 2 Six vendors A B C D E F with the following rates prefix route category preference price 2000 2006 PortaOne Inc All rights reserved www portaone com 36 Porta SIP System Concepts 8610 Cheap 7 0 04 86 Default 5 0 06 86 Cheap 6 0 03 86 Cheap 6 0 025 86 Expensive 5 0 11 8610 Premium 5 0 09 moan oe then when a customer with this routing plan makes a call to 8610234567 the system will arrange the possible routes as follows gt ee The Default route category is first in the route plan A Cheap 7 0 04 This vendor has the highest preference in the Cheap category
111. ing recorded by you congratulating him on successfully activating his VoIP service and giving him other important information If the customer does not answer the phone e g he has connected his SIP adaptor to the Internet but has not connected the phone to it yet and so cannot hear it ringing PortaSIP will try to call him back later Of course after the customer has listened to the message once his first usage flag is reset and no further messages will be played 2000 2006 PortaOne Inc All rights reserved www portaone com 31 Porta SIP System Concepts User authentication In general every incoming call to PortaSIP must be authorized in order to ensure that it comes from a legitimate customer of yours Digest authorization PortaSIP UA ser bZbua ALA 70 68 0 213 216 231 44 34 216 231 44 34 Sipura SPaz000 3 1 5 PortaSIP PortaSIP PortaBilling G gt A7 10l I b INVITE lt i 101 T 100 tr A 101 Ij INVITE A 101 I 401 Unauthor A 101 A ACK A 101 I 401 Unaathor 2 101 85 ACE gt A 102 I INVITE 102 T 100 trying men al A 10Z I INVITE 102 I 100 Trying gt Authorization request gt I lt Auth request accented fa When the first INVITE request arrives from a SIP phone the SIP server replies with 401 Unauthorized and provides the SIP UA with a challenge a long string of randomly g
112. interface in which case the SIP Server can correct the IP address for the RTP stream in SIP messages This method is quite unreliable in some cases it works while in others it fails 3 Use smart UAs or NAT routers or a combination of both which ate able to figure out the correct WAN IP address port for the media by themselves There are several technologies available for this purpose such as STUN UPnP and so on A detailed description of them lies beyond the scope of this document but may easily be found on the Internet Which NAT traversal method is the best There is no ideal solution since all methods have their own advantages and drawbacks However the RTP proxy method is the preferred solution due to the fact that it allows you to provide service regardless of the type of SIP phone and NAT router Thus you can say to customers Take this box and your IP phone will work anywhere in the world In general the smart method will only work if you are both an ISP and ITSP and so provide your customers with both DSL cable routers and SIP phones In this case they can only use the service while on your network NAT Call Scenarios and Setup Guidelines With regard to NAT traversal there are several distinct SIP call scenarios each of which should be handled differently These scenarios differ in that in case 2 the media stream will always pass through one or more NATs as the endpoints cannot
113. ion requests should be sent 1812 by default ACCT PORT Port on the RADIUS server to which accounting requests should be sent value number 1813 by default RAD_TIMEOUT How long the SIP server should wait for a reply from the RADIUS server before retransmit value number 3 by default RAD_RETRIES How many retransmit attempts should be made 2000 2006 PortaOne Inc All rights reserved www portaone com 1 1 9 Porta SIP Administration FAQ Be value number 5 by default Special features configuration meee possible values 0 or 1 FIRSTLOGIN_CLI Appear as CLI AND number on the SIP phone for the first login greeting call value E 164 phone number Send keep alive requests to the caller party originating SIP device possible values 0 or 1 Send keep alive requests to the called party terminating SIP device possible values 0 or 1 SEND START ACCT Send an accounting request to the billing when the call is started this is necessary if you want to display a list of active calls on the billing s web interface possible values 0 of 1 MAX CREDIT TIME Limit maximum call duration for all calls to a specified number of seconds value number 1 means unlimited HUNT_STOP List of SIP error codes which will stop hunting 1 e trying the next route in the sequence value comma separated list of numbers REG EXPIRES MIN Minimal interval between registrations in seconds defaults to 300 This parameter can
114. ipura persists contact support portaone com Provide a full description of the 2000 2006 PortaOne Inc All rights reserved www portaone com 1 1 3 Porta SIP Administration FAQ FAQ problem the ID of the account being used for testing and the relevant parts of the sip log and porta billing log Why can t my debit account initiate 3 way calling using the features of a SIP phone such as Cisco ATA 186 Since 3 way calling requires 2 simultaneous outgoing SIP sessions from one SIP telephone debit accounts will be unable to use it as the first session will lock the account and not allow the second one to go through Therefore if you want to enable your clients to use such services create a credit account for them instead Does PortaSIP support conferencing No Full scale SIP conferencing requires a separate software or hardware solution However you can make use of the features available in some SIP phones such as Cisco ATA 186 to allow your clients to set up simple so called chain conferences For more information please refer to the documentation for each specific SIP phone Can you assist me in integrating SIP device X gateway media server conference server etc made by vendor Y with PortaSIP Yes we can however you will have to purchase an additional consulting contract Generally speaking there should be no compatibility problems between PortaSIP and any standards compliant SIP device However for
115. itate calls from SIP agents behind the firewall sip ua nat symmetric check media src NOTE Cisco GWs are currently unable to log in to the SIP server using the REGISTER method Dial peers Finally create an SIP enabled incoming dial peer dial peer voice 100 voip incoming called number T voice class codec 1 session protocol sipv2 dtmf relay rtp nte l 2000 2006 PortaOne Inc All rights reserved www portaone com 1 0 4 Porta SIP How to Note that this gateway provides no authentication of incoming SIP calls so that potentially anyone could route calls to you from their SIP server This is why the recommended configuration is as follows Call application voice remote 1p tlash app remote authenticate tel dial peer voice 100 voip incoming called number T voice class codec 1 session protocol sipv2 dtmf relay rtp nte application remote ip Thus every incoming call will be authenticated by the IP address of the remote peer Since signaling for the SIP call comes from the SIP server this would be the address of the SIP server This means that calls coming from your own SIP server will be authenticated by billing since your SIP server is entered in the system as a trusted node configure my Cisco gateway to send outgoing calls using SIP Configuration of the Cisco gateway for SIP is not much more difficult than H323 First of all make sure that the rest of your system is configured properly that the
116. ixing time with duration 0 seconds from connect time porta billing pl J l 21 2223 38 07 5 Jul 21 22 38 07 Jul 21 22 38 07 Jul 21 22 38 07 J l 21 22 38 07 66 96 26 134 J l 2L 22138 0785 vendor Mera Systems J l 21 22 38 07 Jul 21 22 38 07 PrepareNexecute Looking up vendor connection Trying to match connection for call Looking for a connection VolP originate Outgoing VoIP env 1 Found connection 306 GetActiveLegIdByAcct matching by the remote IP address Free VoIP Network FVN to Found vendor connection PrepareNexecute InsertActivelLeg The call has not been billed yet but an entry is made in the table of active calls Jul 21 22 38 07 Jul 21 22 38 07 Accounting response is Done The call is terminated stop accounting for one of the call legs arrives Jul 21 22 38 24 NAS IP Address Processing request BE ver1 218 2 2 pid24729 70 68 0 213 User Name 18667478647758 Called Station Id 159044932535 Calling Station Id 18667478647758 Acct Status Type Stop h323 call origin originate h323 call type VOIP h323 setup time h323 connect time h323 disconnect time h323 disconnect cause h323 voice quality h323 conf id call id Acct Session Id 06 37 59 000 GMT Fri Jul 22 20057 06 38 07 000 GMT Fri Jul 22 2005 06 38 24 000 GMT Fri Jul 22 2005 YO ORA 9C25D01D 9ED3C282 83BAC556 8CE06004 l1efdc57
117. k at our standard hourly rate Please also note that a support contract for a non FreeBSD PortaSIP installation will be more expensive and must be negotiated on a case by case basis Please contact sales portaone com for details want to terminate my SIP customers to a vendor that only supports H 323 traffic what should do To do this you need to use a SIP gt H 323 protocol converter Either purchase a dedicated solution available from a number of vendors for instance Mera Networks www mera voip com or use one of your 36xx Cisco gateways with the special IOS feature called IPIPGW In addition to protocol conversion you may also need convert codecs This is not possible with IPIPGW but you can use the Cisco AS53XX gateway by looping one or more paits of E1 T1 ports on it to allow SIP gt ISDN gt H323 call flow Please note that in the latter approach one ongoing session will consume 1 timeslot in each looped E1 T1 2 total as well as 2 DSPs For example if you have two ET interfaces connected back to back the maximum number of simultaneous SIP sessions that you will be able to terminate to your H 323 provider will be 30 and each such session will use 2 DSPs In APPENDIX F Setting up a Back to Back T1 E1 Connection you will find information on how to set up such a back to back connection physically and configure it in Cisco IOS have connected the Cisco AS53XX gateway to PSTN in order to send calls from PSTN to my SIP a
118. l tabs with extra information available on the screen The most important fields are Main form top o Name short name for the customer object this will be used on the web interface O Currency the currency in which this customer will be billed O Opening balance a starting balance for the customer the default is zero o Type isita reseller or retail direct customer Normally most of your customers would be retail customers Only if a customer is reselling your services and you are providing services and billing to his subscribers would he be created as a reseller Address info tab o Email An email address for the distribution of accounting information After the billing period is over a list of CDRs and other statistics will be sent to this address o Bec Blind carbon copy in email may be used for debug and archiving purposes o Summary only Distributes summary only and does not attach a details file might be useful when the amount of calls is very large Additional info tab o Billing period The frequency of accounting information distribution Available billing periods 2000 2006 PortaOne Inc All rights reserved www portaone com 66 Porta SIP Setting up SIP Services Daily one day midnight to midnight sent on the next day Weekly Mon Sun inclusive sent on Monday Bi weekly 1 15 inclusive sent on the 16th and 16 last day inclusive sent on
119. lay name which appears in the caller id for his outgoing calls by default First 2000 2006 PortaOne Inc All rights reserved www portaone com 97 Porta SIP Setting up SIP Services a Name M Name Last Name The phone number assigned to this port The dynamic variable id will be automatically replaced by the account s ID phone number an number Use Auth ID If set to yes the combination Auth ID and Password will be used for SIP authentication Otherwise User ID and Password are used Preferred Codec Select a preferred codec for all calls However the actual codec used in a call will still depend on the outcome of the codec negotiation protocol Possible values G711u G711a G726 16 G726 24 G726 32 G726 40 G729a G723 Use Pref Codec Only use the preferred codec for all calls The call Only will fail if the other endpoint does not support this codec DTMF Tx Method for transmitting DTMF signals to the far Method end Inband Send DTMF using audio path INFO Use the SIP INFO method AVT Send DTMF as AVT events Auto Use Inband or AVT based on outcome of codec negotiation Dial Plan Per line dial plan script see the product manual for a detailed description of the syntax Create an IP phone entry Gi 4 IP Phone Inventory gt Badd cose 0 tout B tog Search cS a Hame Type Managed By MAC Address Description Delete HZ Add IP Phone l
120. le if your SIP customer has account 12021234567 and you want to charge him for incoming calls from PSTN to that number there 2000 2006 PortaOne Inc All rights reserved www portaone com 1 09 Porta SIP How to should be a rate with a prefix matching this number for example 1202 e Inthe product associated with this account add an accessibility entry with this PSTN SIP gateway as the node and the tariff created in the previous step Now calls originating from a SIP phone to 1202 numbers will be charged using the tariff associated in the product s accessibility with the PortaSIP node Calls terminated from the PSTN to the SIP phone will be charged using a different tariff one associated with the PSTN gateway provide error messages from the media server in my users local language First of all you must record a set of all the required voice prompts account_expired cld_blocked and others Convert them into raw format and name the files lt original name gt lt language gt sln for instance the Chinese version of the account expired message will be contained in the file account _expired ch sln Upload the files to the PortaSIP server in the usr local share asterisk sounds directory This will be sufficient to enable the PortaSIP media server to play this voice prompt to SIP phones using 711 GSM and many other popular codecs Unfortunately you cannot perform such online transcoding into the 723
121. located on a public network it can identify the real IP addresses of both parties and correct them in the SIP message if necessary before sending this message further Also the SIP Server can identify the real source ports from which SIP messages arrive and correct these as well This allows SIP signaling to flow freely even if one or both UAs participating in a call are on private networks behind NATS Unfortunately due to the fact that an RTP media stream uses a different UDP port flowing not through the SIP server but directly from one UA to another there is no such simple and universal NAT traversal solution There are 3 ways of dealing with this problem 2000 2006 PortaOne Inc All rights reserved www portaone com 4 4 Porta SIP System Concepts 1 Insert an RTP proxy integrated with the SIP Server into the RTP path The RTP proxy can then perform the same trick for the media stream as the SIP Server does for signaling identify the real source IP address UDP port for each party and use these addresses ports as targets for RTP rather than using the private addresses ports indicated by the UAs This method helps in all cases where properly configured UAs supporting symmetric media are used However it adds another hop in media propagation thus increasing audio delay and possibly decreasing quality due to greater packet loss 2 Assume that the NAT will not change the UDP port when resending an RTP stream from its WAN
122. m 6 9 Porta SIP O Setting up SIP Services Managed by If you want this product to be used for your reseller s accounts so the reseller himself can change the parameters of this tariff and create new accounts using this product choose a customer name from the menu Otherwise choose None here Breakage The left over balance which is considered useless for statistical purposes Accounts with a balance below the breakage will be counted as depleted Maintenance period The surcharge application interval which will be reflected in call history as a separate line each time it is charged at the end of a specified period Maintenance fee the surcharge amount Info URL If you have an external server with a description of product features enter the URL here Your customers will be able go here from their self care page Description your comments about the intended use of this product 3 Click l Save 4 Click on the Accessibility tab to edit this product s accessibility Enter Node and Tariff into the product s accessibility list The Accessibility List has two functions it defines permitted access points nodes and access numbers and specifies which tariff should be used for billing in each of these points 1 When the Accessibility tab is selected click on the El Add icon In the accessibility entry window select the node where your IVR is running and choose the appropriate tariff
123. mended default settings are RTPPLocalPolicy nat RTPPRemotePolicy direct This feature which was not available in previous versions of PortaSIP provides an improved way of controlling RTP proxying For SIP to SIP calls the RTP proxy will be activated only when one of the SIP phones is behind a firewall so if both SIP phones are on public IPs they will be allowed to communicate directly e Ifyou plan to use the first login greeting feature please make sure you prepare your voice prompts convert them to the different codecs which will be utilized by your users e g 723 and upload them to the PortaSIP server 2000 2006 PortaOne Inc All rights reserved www portaone com 7 Porta SIP System Concepts 1 System Concepts 2000 2006 PortaOne Inc All rights reserved www portaone com g Porta SIP System Concepts PortaSIP s Role in Your VoIP Network NAT Mes 3 SIP Server ITSP s B2BU A ork Router rR SIP phone SoftPhone a NAT gt S mP ee Radi eee Billing Engine Phone ATA186 Router D Porta SIP SoftPhone PortaSIP is a call control software package enabling service providers to build scalable reliable VoIP networks Based on the Session Initiation Protocol SIP PortaSIP provides a full array of call routing capabilities to maximize performance for both small and large packet voice networks PortaSIP allows IP Telephony Service Providers to deliver
124. mmon the wizard which will help you construct the correct period definition Click Help for more information on period format definition If you do not differentiate between peak and off peak rates just leave this field empty Off Peak Description a description of the off peak period automatically filled in by the off peak period wizard thus you do not have to fill in this field Destination group set if you wish to enter rates in the tariff not for every individual prefix but for a whole group of prefixes at once you should create a destination group set and destination groups beforehand Leave this select menu empty for now Free seconds The number of free seconds allowed for each call In order to claim free seconds the length of the call must be 2000 2006 PortaOne Inc All rights reserved www portaone com 56 Porta SIP Setting up SIP Services at least one billing unit first interval see the Enter Rates section above Oo Post Call Surcharge percentage of the amount charged for the call Oo Login Fee amount to be charged immediately after the first user authentication 1 e after the user enters his PIN o Connect Fee amount to be charged for each connected call call with a non zero duration o Round charged amount Instead of calculating CDRs with a 5 decimal place precision round up CDR amount values e g to cents so that 1 16730 becomes 1 17 Set the rounding pattern to XXXX
125. n described in the paragraph Tips provide information that might help you solve a problem 2000 2006 PortaOne Inc All rights reserved www portaone com 4 Porta SIP PortaSIP User Guide Hardware and Software Requirements Server System Recommendations One UNIX Server A minimum of 40 GB of available disk space this space is required for storing various log files A processor running at 2 4 GHz or greater Additional processor speed is needed for networks with a high call volume At least 512MB of RAM RDRAM or DDR 1 GB recommended At least one USB port For information about whether particular hardware is supported by FreeBSD from the JumpStart Installation CD consult the related document on the FreeBSD website http www freebsd org doc en_ US 1 SO8859 1 books faq hardware html Client System Recommendations OS Windows 95 XP UNIX or Mac OS Browser Internet Explorer 6 0 or higher Netscape 7 1 Mozilla 1 7 ot higher supporting DOM and with JavaScript enabled Spreadsheet processor MS Excel Display settings o Minimum screen resolution 1024 x 768 O Color palette 16 bit color minimum NOTE To view downloaded CSV Comma Separated Values files in Windows please do the following to match PortaBilling s default list separator My Computer gt Control Panel gt Regional Settings gt Number gt List Separator type 2000 2006 PortaOne Inc All rights res
126. nclude users of Exchange Instant Messaging Signin name Advanced Cancel Help 2000 2006 PortaOne Inc All rights reserved www portaone com 1 9 g Porta SIP Appendices 3 Click the Configure settings radio button and enter the Server name of IP address using either the IP address of the PortaSIP server or its name in DNS Make sure that the UDP radio button is selected then click OK SIP Communications Service Connection Configuration Select which method should be used to configure pour connection to communications Service Automatic configuration Server name or F address demo portaone com Connect using TCP f TLS UDP Cancel Help 4 Sign out and then sign in again You should see the pop up dialog below Fill it in as follows Sign in name in the form usernameQaddress where username is the name of the appropriate account in PB and address is either the IP address of the PortaSIP server or its name in DNS Enter the name of the appropriate PB account as the User Name and the appropriate account password as the Password then click OK You should now see your status change to online Sign In to a SIP Communications Service Enter your sign in name user name and password to sign in ta demo portaone com Sign in name fo TSQ00 derno portaone com Example zomeonemesample com User name iar 18000 Examples domain usern
127. ncludes the phone prefix and country name The country subdivision is optional You can use the Description column to store extra information about the destination for example if it is a mobile or fixed number E A Destinations fe TE ada E Save close Download Get defaut set upio Logout Bio Prefix country Description gt ABCDEFGHIJKLMNOPQRSTUYWXYZ Edit Prefix Country Subdivision Description Delete 1778 Not Applicable vw Not Applicable Mobile 4 Click Save 5 Repeat these steps for any additional destinations you would like to add Uploading a set of destinations from a file 1 In the Management section of Admin Index choose Destinations 2 Click on Default set to download a set of destinations as a CSV Comma Separated Values file Es File Edit View Insert Format Tools Data Window Help 18 2 7 24 Prefix Country tw Description 93 AF Proper 93229 AF Kabul 9 32E 08 AF Operator 93321 AF Kandahar 9344000 AF Herat City 9344001 AF Herat City 9344002 AF Herat City 935051 AF Mazar l Sharif 93702 AF Mobile 93708 AF Mobile 355 AL Proper 35536 AL Mobile 3554 AL Tirana 35542 AL Tirana 35566 AL Mobile 35569 AL Mobile 213 DZ Proper 2132 DZ Algiers 2136 DZ Mobile 3 Open this file in Microsoft Excel or any other suitable program Edit the data if necessary 4 Save the file and close it in Excel 5 Switch back to the PortaBilling web interfa
128. next billing unit in seconds Price First per minute price for first interval Price Next per minute price for next interval Off peak Interval First first billing unit in seconds for off peak time Off peak Interval Next next billing unit in seconds for off peak time O Off peak Price First per minute price for first interval for off peak time O Off peak Price Next per minute price for next interval for off peak time OO 0 0 O NOTE Off peak fields appear only if an off peak period has been defined for the tariff o Formula launches the wizard for creating a custom rating formula o Effective from If you want this rate to take effect sometime in the future you can either type in a date manually or use the calendar click on the DD MM YYYY link 2000 2006 PortaOne Inc All rights reserved www portaone com 5 g Porta SIP Setting up SIP Services NOTE When using the calendar you can specify that the date you are entering is ina different time zone than your present one PortaBilling will then automatically adjust the time o The Hidden Forbidden or Discontinued flags are optional 4 Click the lal Save button in the toolbar or the icon on the left side of the row 5 Repeat these steps if you need to enter more rates Tariffs with routing extensions These tariffs are created for your vendors In addition to the billing parameters described above you can also specify
129. ng can automatically create a configuration file for the SIP phone and place it on the provisioning server The only configuration setting which is required on the IP phone side is the address of the provisioning server i e where it should send a request for its configuration file When the IP phone connects to the Internet it will retrieve a specific configuration file for its MAC address from the TETP or HTTP server and adjust its internal configuration If you decide later to change the address of the SIP server you need only update it once in the profile and new configuration files will be built for all user agents Each user agent will then retrieve this file the next time it goes online 2000 2006 PortaOne Inc All rights reserved www portaone com 49 Porta SIP System Concepts Porta Billing Provisioning server Account phone line p E passug IP phone inventory record MAC address ters IP phone profile General parame Request for provisioning information gt Configuration fle gt IP Phone IP Phone The config file is specific to each user agent as it contains information such as username and password thus the user agent must retrieve its own designated config file The following are defined in the billing configuration e The IP phone profile so that the system knows which generic properties e g preferred codec to place in the configuration file e An entry
130. nnections oi Eton 6 LA nee Vendor Name amp lobalNet j Opening Balance 0 00000 USD Balance 0 00000 USD Address Info Additional Info User Interface Accounts Notepad dit Hame Login Password Delete GlobalNet S1P demovendor demo123 x El 6 6 E 4 Edit Vendor gt E add i save fl savetciose close cons 2 connections MB Logout B tog Vendor Name amp lobalNet Bi 7 Opening Balance 0 00000 USD Balance 0 00000 USD Edit Hame Login Password Delete GlobalNet SIP demoVendor ee x 6 Click Close in order to return to the Vendors admin page 7 Repeat steps 2 6 to add all of your vendors Define Connections Connections are points at which calls leave or enter a network and are directed to or from vendors whereby costing occurs 1 In the Management section of the admin interface choose Vendors 2 Click on the Connections icon next to the vendor name E 4 Vendor Management gt E add close B toot B tog Search CDRs Hame Connections Currency Balance E mail Delete GlobalNet USD 0 00000 E Mes A h PSTN from Yendor YolIP from Yendor PSTN to Yendor YoIP to Yendor Remote IP Transl Rule Outgoing Rule CLI Transl Rule Account Edit Load H323 SIP z Delete Tariff Description Capacity 1213 56 70 2 AE I FA cicbainet s1r v in o E SIP Phone subscribers WY Termination 30
131. number formats is to choose a certain number format as the standard for your network and make sure that calls travel on your network only in this format The ideal candidate for such a format is E 164 of course it is highly recommended that you use this 2000 2006 PortaOne Inc All rights reserved www portaone com 37 Porta SIP System Concepts same format in billing as well When a call arrives from your customer with a phone number from a customer specific number plan PortaSwitch will convert the number into your network format It will then travel on your network until it is sent to a vendor for termination Just before this happens it can be converted into the vendor specific format Customer based translation rules Very often your customer will have his own numbering format for example dialing 00 for international numbers while dialing just the phone number for local calls Customer based translation rules allow you to convert a number from a format specific to this particular customer There is a special dialing rules wizard available to make such configuration easier so that customers can even do this themselves Customer based translation rules have two applications e When a number is submitted for authorization these rules will be applied with the resulting number used to search rates Thus if your customer dials 0042021234567 you can convert it to 42021234567 and find the correct rates for the 420 prefix e
132. obvious reasons we only provide detailed setup instructions for the Cisco AS5300 gateway Can use PortaSIP with a billing system other than PortaBilling100 Yes this is possible PortaSIP uses the standard Radius protocol to communicate with the billing engine and its AAA behavior was purposely made very similar to that of Cisco IOS So it should work with any billing system that supports Radius and can bill Cisco gateways However advanced services such as billing assisted routing abbreviated dialing PortaUM integration and so on require support from the billing engine Detailed specifications of the protocol used to exchange information between PortaBilling100 and PortaSIP are available upon request 2000 2006 PortaOne Inc All rights reserved www portaone com 1 1 4 Porta SIP Administration FAQ Can PortaSIP be installed on a different Unix like operating system Linux Solaris etc than FreeBSD Yes this is possible PortaSIP is easily portable to any modern Unix like operating system However the base price of the product only covers PortaSIP FreeBSD installation to a clean server using the CD provided by Porta Once you have purchased the product we can provide you with the full source code used to build PortaSIP so that you can compile and install it yourself to an operating system of your choice Alternatively we can do this for you however you will be additionally charged for the time required for this tas
133. of the values for dynamic variables G A IP Phone Profiles lel Save el Save amp Close amp Close PE Logout gt U Save fel Saveaclose Close zoo Name Sipura standard j Effective From Type Sipura 2000 Date immediately YYYY MM DD Description Standard profile for residential SIP Time HH24 MI 55 Managed By Administrator only Discontinued Proxy sip mydomain com Outbound Proxy Use Outbound Proxy yo Y Register Yes V Register Expires 3600 Display Name firstname midinit flastname UserID Auth ID Use Auth ID Preferred Codec Use Pref Codec Only DTMF Tx Method Auto v Dial Plan xx 3469 11 0 00 2 9 xxxxxx 1xxx 2 9 xxxxxxS0 xxxxxxxxxxxx In the Line 1 and Line 2 tabs you can specify parameters for both phone ports of your IP phone calls from the SIP registration server Use Outbound If set to no the SIP server defined by the Proxy Proxy parameter will be used for all registrations and outgoing calls Register Whether or not the IP phone should register with the SIP server this is required to receive incoming calls Register Expires Registration lifetime in seconds the Expires value in a REGISTER request The IP phone will periodically renew registration shortly before the current registration expires Display Name The subscriber s disp
134. off net on net etc is authorized authenticated and billed The system is also able to provide Debit billing i e to disconnect a call if the account balance falls below zero Also BZ2BUA can automatically disconnect the other call leg if the SIP phone goes offline due to a network problem e RTP Proxy The RTP Proxy is an optional component used to ensure a proper media stream flow from one SIP telephone to another when one or both of them are behind a NAT firewall e Media Server The Media Server is used to play a number of short voice prompts to an SIP user when an error occurs such as zero balance invalid password and so on 2000 2006 PortaOne Inc All rights reserved www portaone com 1 1 Porta SIP System Concepts Call Process Supported Services SIP UA lt gt SIP UA An example a customer purchases our VoIP services and two of his employees A and B are assigned SIP phone numbers 12027810003 and 12027810009 respectively For convenience the administrator creates two abbreviated dialing rules 120 for 12027810003 and 121 for 12027810009 Also he sets up standard US dialing rules so that users can dial local numbers in the 202 area code by just dialing a 7 digit phone number When the called party is online Porta 4 Billing A A SIP phone A SIP phone B This is the simplest case e User A dials user B s number 121 His SIP user agent sends an INVITE request to the SIP server 1 e The
135. oicemail box PortaUM required if the called number belongs to an account in PortaBilling but this account is not currently registered to the SIP server is offline e Route the call to one of the gateways for termination according to the routing rules specified in PortaBilling Routing of SIP on net calls The SIP server automatically maintains information about all currently registered SIP user agents so it is able to determine whether a call should be sent directly to a SIP user agent Routing of off net calls You can have different vendors for terminating off net calls For example you can terminate calls to the US either to AT amp T via a T1 connected to your gateway in New York or to a remote gateway from Qwest 2000 2006 PortaOne Inc All rights reserved www portaone com 33 Porta SIP System Concepts Rate routing parameters Ordinarily tariffs define the termination costs for each connection to a vendor If you create a tariff with the Routing type a few mote fields will be added to rates in that tariff e Route category you can split this into categories such as Premium Cheap etc and use these categories in routing plans e Preference routing priority 0 10 higher values mean higher priority 0 means do not use this rate for routing at all e Huntstop signals that no routes with a lower preference should be considered This allows you to easily manage both termination cos
136. ol specific format e g 0x00150015 You may find more information at http www cisco com en US products hw gatecont ps514 prod ucts_configuration_example09186a00800c3a50 shtml However this complicated way of entering the parameters makes it virtually impossible for end users to employ Address htp 192 168 0 16 dev AE Line 1192 168 0 16 sk 255 255 255 0 a e SS a O demo portaone com demo portaone com CS 0000 CE EL ne DOO Momo OOO O ss 16364 demo paraone com emo portaone cam demo portaone com fo ooo000te DOUISUUTS Fortunately PortaBilling provides a safe and user friendly way to configure your Cisco ATA from the web interface via Cisco ATA Expett 1 In the Management section of the Admin Index page choose Cisco ATA Expert 2 Type in your Cisco ATA IP address as well as the administrator s passwotd if you have set up one E 4 Cisco ATA Expert Login EEE i tagout EE IP 195 140 247 242 UiPasswor o O NOTE The PortaBilling ATA Expert needs to communicate directly with the Cisco ATA So make sure that the ATA is connected to the network and configured with an IP address This IP address must be either a public IP address accessible from anywhere on the Internet or a private IP address e g 192 168 xxx xxx which is accessible from the PortaBilling web server 2000 2006 PortaOne Inc All rights reserved www portaone com 19 Porta SIP Set
137. onality in its entirety or completely correctly So theoretically STUN may be used in conjunction with PortaSIP s RTP proxy if a phone detects that it can bypass NAT via STUN it will act as if it were on a public IP address and the RTP proxy will not be engaged Unfortunately in practice activating STUN only makes matters worse due to flaws in STUN implementation for IP phones Using two different approaches to handling NAT concurrently is the same as adding flavorings salt pepper etc to a stew by following several recipes from different cookbooks at the same time even a slight mix up will probably result in your adding some of the seasonings twice while not putting others in at all and the result will be something which no one can eat Currently one very common problem situation is that where a SIP phone is behind a symmetric NAT and obtains its public IP address from STUN putting this into the contact information This confuses the RTP proxy since PortaSIP regards the SIP phone as being on a public IP address so that no RTP proxy is used the result is one way audio So the simplest answer is yes You can use STUN to avoid usage of an RTP proxy in some cases At the present moment however due to unreliable STUN support on the IP phone side the safest option is to avoid using STUN 2000 2006 PortaOne Inc All rights reserved www portaone com 1 1 7 Porta SIP Administration FAQ PortaSIP configuration Po
138. ot path for devices which only have TETP protocol support edit the etc inetd conf file and the s inline parameter for tftpd server startup Do not forget to restart inetd afterwards 2000 2006 PortaOne Inc All rights reserved www portaone com 1 02 Porta SIP How to 2000 2006 PortaOne Inc All rights reserved www portaone com 1 03 Porta SIP How to configure my Cisco gateway to accept incoming SIP calls and terminate them to a telephony network Configuration of the Cisco gateway for SIP is not much more difficult than H323 First of all make sure that the rest of your system is configured properly that the gateway can place the outgoing calls and is able to communicate with the billing using RADIUS Codecs First of all make sure you have set up a list of codecs which are supported by your SIP agents on your GW Your actual configuration might differ but here is a good example which should work in most cases voice class codec 1 codec preference 1 g723r63 codec preference 2 g 29r8 codec preference 3 g7 29br8 codec preference 4 g723r53 codec preference 7 g726r16 codec preference 8 g7 26r24 codec preference 9 g726r32 codec preference 10 g llalaw codec preference 11 g llulaw codec preference 12 g 23ar53 codec preference 13 g 23ar63 SIP agent Now enable the SIP agent functionality on your gateway Also enable it on gateways where NAT symmetric traversal is supported as this will facil
139. ough this is rarely done with SIP service accounts O Opening balance the initial balance on the account Account Info tab O O Account type account type select credit for post paid and debit for prepaid service Credit limit For a credit account specify the credit limit If you leave this field empty it means there is no credit limit for this account but a customer credit limit may still apply VoIP password This password is used for SIP services as well The account ID and this password will be used to authenticate SIP server login Email Enter the account owner s email address here If he ever forgets his password for the web self care pages he will be able to reset it and a new password will be sent to this email address You can also just leave this field empty Batch A batch is a management unit for accounts The batch name is alphanumeric You can type a new name here or use an existing name in order to generate more accounts for the same batch 2000 2006 PortaOne Inc All rights reserved www portaone com 6 g Porta SIP Setting up SIP Services Additional Info tab Oo Preferred Language This is a custom attribute which is transferred to the IVR Leave English here if you are unsure whether your IVR supports this function o Redirect Number redirect number discussed in the Advanced features section leave this empty since it is not used by PortaSIP o UM Enabled c
140. ountBalance Jul 21 22 38 24 Charging account s owner for the call Jul 21 22 38 24 Charging customer 2 PortaOne 0 0803 Jul 21 22 38 24 PrepareNexecute UpdateCustomerBalance Jul 21 22 38 24 Charging vendor for the call Jul 21 22 38 24 Charging vendor 152 Mera Systems 0 0475 Jul 21 22 38 24 Inserting CDR Jul 21 22 38 24 PrepareNexecute InsertVendorCDR Jul 21 22 38 24 PrepareNexecute UpdateVendorBalance Jul 21 22 38 24 Accounting response 2000 2006 PortaOne Inc All rights reserved www portaone com 90 Porta SIP Jul 21 22 38 24 Setting up SIP Services Done Accounting for the second incoming call leg arrives Jul 21 22 38 24 NAS IP Address User Name Called Station Id Calling Station Id Acct Status Type h323 call origin h323 call type h323 setup time h323 connect time h323 disconnect time h323 disconnect cause h323 voice quality h323 conf id call id Acct Session Id Processing request BE ver1 218 2 2 pid24729 70 68 0 213 18667478647758 380443333333 18667478647758 Stop answer VoIP 06 37226 000 GMT Fri Jul 22 06 38 06 000 GMT Fri Jul 22 06 38 23 000 GMT Fri Jul 22 YO YO 9C25D01D 9ED3C282 83BAC556 8CEO600A Jefdc57b 80c3c2c 192 168 1 180 Jefdc57b 80c3c2cQ 192 168 1 180 2005 2005 2005 I I 2000 2006 PortaOne Inc All rights reserve
141. p of the list thus becoming the first route i 4 Test Dialplan fe e ase La omes i toot Protocol 4 Date and Time H323 SIP YYYY MM DD HH Mi 8610234567 Default OM Csearh Phone Humber Routing Plan Destination Country Description Price Route Category Preference Huntstop Route to Vendor Connection Tariff 86 CHINA Proper 0 02500 USD Default 5 N 45 12 156 200 YendorD Termination to vendor D Yendor D 86 CHINA Proper 0 03000 USD Default N ny gw 01 vendor C Termination to carrier Vendor C CHINA Beijing 0 04000 USD Default N ny gw 01 vendor A Termination to carrier A Vendor A 86 CHINA Proper 0 06000 USD Default N CHINA Beijing 0 09000 USD Default N 86 CHINA Proper 0 11000 USD Default N ny gw 01 vendor B Termination to carrier B Vendor B 64 67 2191 VendorF Termination to vendor F Vendor F 193 50 123 6 VendorE Termination to carrier E Vendor E i 4 Test Dialplan ES EEE AE CT EE Protocol 1 Date and Time H323 SIP YYYY MM DD HH Mi 861045676 Default o Phone Humber Routing Plan Destination Country Description Price Route Category Preference Huntstop Route to Vendor Connection Tariff 1 8610 CHINA Beijing 0 04000 USD Default 6 ny gw 01 vendor A Termination to carrier A vendor A 2 86 CHINA Proper 0 03000 USD Default ny gw 01 vendor C Termination to carrier C Vendor C 3 86 CHINA Proper 0 06000 USD Default ny gw 01 vendor B Termination to carrier B Vendor B 4 CHINA Be
142. protocol sipv2 h323 ivr out PortaBilling Routing SIP h323 ivr out PortaBilling Notify NAT 21 Jul 23 37 26 lefdc5 b 80c3c2c 192 168 1 180 b2bua AAA request accepted processing response Billing authorizes the call and provides information about call routing 3 possible routes are returned Cisco AVPair h323 ivr in Portabilling Routing 3804433333330195 234 212 1 expires 300 credit time 1 Cisco AVPair h323 ivr in PortaBilling Routing 380443333333006 96 26 134 500lexpires 300 7 credit time 1 Cisco AVPair h323 ivr in PortaBilling Routing 380443333333 67 105 130 102 auth PortaSoftware xxxxx expires 300 credit time 1 h323 billing model h323 billing model 0 Cisco AVPair h323 ivr in Tariff Porta SIP Cisco AVPair h323 ivr in PortaBillang CompleteNumber 360443333333 h323 return code h323 return code 13 h323 currency h323 preferred lang h323 currency USD h323 preferred lang en RTP proxy is engaged to send the call via the first route 21 Jul 23 37 26 GLOBAL rtpproxy received command U lefdc57b 80c3c2c 192 168 1 180 62 244 32 30 16384 e97ded4c2b244b0cb4007cedb742f088 543650df2972e81b25e307d7ed2a2e04 21 Jul 23 37 26 GLOBAL rtpproxy new session lefdc57b 80c3c2c 192 168 1 180 tag e97ded4c2b244b0cb4007cedb742f088 medianum 0 requested type strong 21 Jul 23 37 26 1lefdc57b 80c3c2c 192 168 1 180 rtpproxy new session on a por
143. pstnzsip tel call application voice pstn2sip authenticate by dnis call application voice pstn2sip skip password yes call application voice pstn2sip authorize yes Call application voice pstnZsip dial account id yes dial peer voice 100 pots incoming called number T application pstn2sip volice port Od I The example above is for when you receive incoming calls with phone numbers already in E 164 If the number is received in a local format you will have to use the translate feature in the PSTN2SIP script to convert the number into E 164 For instance if you receive a US phone number in NANP area code phone number you should add the following command to the application configuration call application voice pstn2sip translate 1 Then configure your gateway to send outgoing calls to the SIP server according to the instructions in the previous topic SUpport incoming H323 and SIP calls on the same gateway This configuration is supported as Cisco GW can handle both H323 and SIP calls at the same time However please note that Cisco matches an incoming dial peer by the incoming called number not by the protocol Thus the dial peer shown below will match both incoming SIP and H323 calls even if it gives the session protocol sipv2 dial peer voice 101 voip description Incoming SIP calls incoming called number 2000 2006 PortaOne Inc All rights reserved www portaone com 1 06 Porta SIP How to
144. r data using the PortaBilling default format The rates table may be prepared using a spreadsheet processor i e Microsoft Excel and easily imported into PortaBilling This is very convenient if you are going to make many changes For example you might increase all prices by 10 2000 2006 PortaOne Inc All rights reserved www portaone com 59 Porta SIP Setting up SIP Services 4 If you are not in Tariff Management for your tariff go to the main menu click on Tariffs and then click on the tariff name In the Edit Tariff window move the mouse over the Download button and hold it there until a popup menu appears Choose the Now menu item and click on it This will download the current set of rates empty but will also provide you with an overview of the file structure You will see the File download dialog and be prompted to choose whether to save the file or open it from the current location We recommend that you save the file into the folder you will be using in the future to store tariff data files then open it in Excel Now you should see something similar to the screenshot below ia File Edit View Insert Format Tools Data Window Help X Clasa B C D E E G H J K L M N o P_ Name _ICurrency Descriptior Short Dest Off Peak Description 7 SIP Phone USD This tariff SIP phone PERIOD From 20 00 until 06 00 Off peak P Destinatior Free Seco Post Call Login Fee Connect Fee startstop hr 20 5 0
145. rd for your IP phone in the IP phone inventory making sure to enter the correct MAC address Assigning a phone number to the IP phone Make sure your SIP service is provisioned according to the instructions given above in this guide Create a new account with the product you allocated for the SIP service with an account ID identical to the phone number Assign this account for provisioning on a certain IP phone IP phone settings Connect the IP phone to the Internet If the phone has not been pre configured for your provisioning server by the vendor enter provisioning information into the phone manually Wait until the configuration files are updated on the provisioning server Testing After your phone downloads the configuration from the server and successfully registers on the SIP server for the first time you will receive a first login greeting call Make a test phone call 2000 2006 PortaOne Inc All rights reserved www portaone com g 4 Porta SIP Setting up SIP Services Create an IP phone profile i 4 IP Phone Profiles gt 5 ada cose Logout B Los Effective From Hame Managed By wos E La v ar Hame Type Effective From Managed By Discontinued Description Delete HZ Add IP Phone Profile Name Managed By Type As Copy Of Effective From Date immediately yvyy MM DD Time 00 00 00 HH24 MI 55 1 Inthe Man
146. ri Jul 22 2005 h323 disconnect cause 66 h323 conf id 9C25D01D 9ED3C282 83BAC556 8CEO600A Cal l id Jefdc57b 80c3c2c 192 168 1 180 Acct Session Id lefdcs b 80c3c2c 192 168 1 180 Acct Session Time 0 Acct Delay Time 0 h323 remote address 195 234 212 1 h323 session protocol sipv2 NAS Port Id 5060 Exec Program Log porta billing pl Jul 21 22 37 59 This call belongs to the environment 1 pb Jul 21 22 37 59 h323 conf id 9C25D01D 9ED3C282 83BAC556 8CE0600A 1 call id lefdc57b 80c3c2c 192 168 1 180 1 Jul 21 22 37 59 Found a call in cache with such id Billing re uses information in the call cache to speed up account info lookup Jul 21 22 37 59 Copied account 18667478647758 147059 credit balance 0 55021 limit none of customer PortaOne 2 balance 9996029 96819 limit 0 00000 from 70 68 0 213 into the current request Jul 21 22 37 59 PrepareNexecute GetActiveLegIdByAcct Jul 21 22 37 59 End of the outgoing failed call for logged in account Waiting another outgoing call or hang up Jul 21 22 37 59 Looking up vendor connection Jul 21 22 37 59 Trying to match connection for call Jul 21 22 37 59 Looking for connection VolP originate Jul 21 22 37 59 Outgoing VoIP matching by the remote IP address 195 234 212 1 env 1 Jul 21 22 37 59 Call goes to our trusted node mera Jul 21 22 37 59 Connection to vendor not found
147. rst keep alive at the very beginning of the call when the SIP phone should presumably be online then it assumes that the SIP UA does not 2000 2006 PortaOne Inc All rights reserved www portaone com 1 1 6 Porta SIP Administration FAQ support this functionality and disables keep alives for this session In any case it is recommended to choose a SIP UA which supports re INVITEs e g Sipura do not want to use an RTP proxy since it will increase the amount of required bandwidth can I use STUN instead The STUN REC http www faqs org rfcs rfc3489 html states This protocol is not a cure all for the problems associated with NAT STUN is merely a service that can be installed on a server such as PortaSIP allowing a STUN enabled SIP phone to communicate with it and detect the type of firewall it is behind and the public IP address of the NAT router Thus a SIP phone may obtain certain information by communicating with a STUN server but this will not have any effect on the way NAT handles IP packets traveling to or from the phone In the case of a cone firewall STUN information may help the SIP phone to determine in advance which IP address and port the remote party can use to communicate with it However in the case of a symmetric NAT this will not work and so an RTP proxy is still required Moreover since this is a relatively new technology many phone vendors have not implemented the STUN functi
148. rta 4 Billing Server 1 Porta K4 SIP Server 2 Frankfurt SIP phone A SIP phone B In the example above user A assigned SIP phone number 12027810003 and registered to PortaSIP in New York calls user B with phone number 4981234567 who is currently registered to PortaSIP in Frankfurt e A dials B s number 4981234567 His SIP user agent sends an INVITE request to PortaSIP server 1 1 e The SIP server sends an authorization request to the billing 2 e After all the usual authorization checks the billing discovers that the dialed number is one of our SIP accounts but is currently registered to PortaSIP server 2 It instructs the SIP server to route this call to the IP address of PortaSIP 2 3 e PortaSIP server 1 sends an INVITE request to PortaSIP server 2 4 e Upon receiving this INVITE PortaSIP 2 sends an authorization request to the billing 5 e The billing authorizes the call since it comes from a trusted node and requests that the call be sent to the locally registered SIP UA 6 e The SIP server sends an INVITE request to the SIP phone 7 2000 2006 PortaOne Inc All rights reserved www portaone com 1 4 Porta SIP System Concepts SIP UA gt PSTN Porta Billing SIP phone A GW NY 02 Phone C 12 34 56 78 e User A attempts to call his co worker user C C has not been assigned a SIP phone yet thus he only has a normal PSTN phone number from the 202 area code and A dial
149. rtaSIP provides a unified configuration tool Even if a system consists of several components using different technologies and configuration methods you just have to edit one simple configuration file This master configuration file is then used by PortaOne configuration scripts to manage and provision other modules e g SER B2BUA and so on porta sip conf This is the only file you need to edit in order to modify PortaSIP parameters Every row starting with is considered to be a comment the other lines will contain VAR VALUE pairs separated by a colon 6 A This file is created automatically during installation Thus assuming you provided correct parameters during installation you do not have to change anything General configuration IP address of the SIP environment SIP_PORT Port on the SIP server which SIP phones should connect to value number default 5060 CANONIC_NAME Fully qualified domain name for this SIP server so your customers can use contact information in the form 1234 sip domain com I ENV PortaBilline virtual environment id for this SIP instance note that this is a numeric ID 1_env and not the environment name use the porta admin pl utility on the slave server to find the correct value RADIUS configuration PB MASTER IP address of the PortaBilling100 master host RADIUS secret key for RADIUS requests to the billing value string AUTH PORT Port on the RADIUS server to which authenticat
150. rts of an incoming INVITE request if a certain pattern matches then IP authentication will be used PortaSIP may initiate IP authentication if any of the following match a pattern User Agent SIP header Remote IP address the address from which the INVITE request is received 2000 2006 PortaOne Inc All rights reserved www portaone com 39 Porta SIP System Concepts e Any of the SDP fields By default the following SIP UAs are considered incapable of digest authentication so that IP authentication is applied e Cisco VoIP gateway any Cisco gateway running IOS this does not apply to Cisco ATA 186 188 e Nextone SBC e Sonus switch e Mera SIP HIT e Quintum gateway e Asterisk gateway Please ask the PortaOne support team for assistance in adjusting the information in this table to reflect the desired configuration of your network Understanding SIP Call Routing When the PortaSIP server has to establish an outgoing call it must find out where the call is being sent to To do this it will ask billing for a list of possible routes In this case the routing configuration is in one central location and billing can use information about termination costs to choose the best route least cost routing When a call goes through the PortaSIP server the SIP server may e Direct the call to one of the registered SIP clients if the called number belongs to the registered agent e Optionally direct the call to the v
151. rved www portaone com 99 Porta SIP System Concepts Service announcements via the media server A customer might be unable to make a call not only due to network problems but also for various administrative reasons for example if his account is blocked or he does not have enough money on his account If the end user can be informed of such administrative problems instead of just being given a busy signal this will greatly simplify troubleshooting Here is what would happen in the event that for instance an account which is blocked attempts to make a call e The customer tries to make a call SIP proxy receives the INVITE request and sends an authorization request to the billing e PortaBilling determines that this account is blocked An authorization reject is returned to the SIP server In addition to the h323 return code a special attribute is sent back to the SIP server This attribute contains a description of the type of error in this case user_denied e The SIP server receives the authorization reject from the billing However instead of just dropping the call it redirects the call to the media server including the error message as a parameter e The media server establishes a connection with the SIP UA It locates a voice prompt file based on the error type and plays it to the user After this the call is disconnected The media server and prompt files are located on the SIP server So as to avoid dynam
152. s 3001234 A s SIP user agent sends an INVITE request to the SIP server 1 e The SIP server sends an authorization request to the billing 2 e Billing performs several operations O Checks that such an account exists that it is not blocked expired that the supplied password is correct that the account is allowed to use SIP services etc o Performs a dialed number translation according to the customer dialing rules or abbreviated dialing table so 3001234 will be converted into 12023001234 O Checks if A is actually allowed to call that number and what is the maximum allowed call duration o Discovers that the destination number is off net O Computes the routing for this call to the external vendors according to their cost and preferences and the customer s routing plan Based on the results of the above operations billing sends an authorization response to the SIP server 3 e The SIP server tries to send a call to all routes returned by the billing sequentially until either a connection is made or the list of routes is exhausted 4 e When the call is finished the SIP server sends accounting information to the billing 2000 2006 PortaOne Inc All rights reserved www portaone com 1 5 Porta SIP System Concepts Terminating SIP calls to a vendor using VoIP Porta 4 Billing SIP phone A Phone C e An example we are able to terminate calls to the US and Canada to a vendor X Telecom This woul
153. s behavior and may change the default policy for instance activating the RTP proxy even for calls when both phones are on public IP addresses See PortaSTP related parameters in porta billing conf for details on RTP proxy policy configuration PortaSIP cannot detect whether a remote gateway supports Comedia extensions symmetric NAT traversal So if you do not use your own gateway for termination you should clarify this matter with your vendor and set up your RTP policy for off net calls accordingly Auto provisioning of IP phones If you provide your VoIP customers with IP phone equipment you know how laborious and yet important the task of performing initial configuration is If the equipment is not configured properly it will not work after being delivered to the customer Or even if it works initially problems will arise if you need to change the IP address of the SIP server How can you reconfigure thousands of devices that are already on the customer s premises There are two ways to manage the device configuration Manual provisioning The administrator must login to the device provisioning interface typically HTTP and change the required parameters There are several drawbacks to this method e The IP phone must be connected to the Internet when the administrator is performing this operation e The administrator must know the device s IP address e The IP phone must be on the same LAN as the administrator
154. s to a SIP based phone call The first is the signaling that is the protocol messages that set up the phone call and the second is the actual media stream i e the RTP packets that travel directly between the end devices for example between client and gateway SIP Signaling SIP signaling can traverse NAT in a fairly straightforward way since there is usually one proxy The first hop from NAT receives the SIP messages from the client via the NAT and then returns messages to the same location The proxy needs to return SIP packets to the same port it received them from 1 e to the IP port that the packets were sent from not to any standard SIP port e g 5060 SIP has tags which tell the proxy to do this The received tag tells the proxy to return a packet to a specific IP and the rport tag contains the port to return it to Note that SIP signaling should be able to traverse any type of NAT as long as the proxy returns SIP messages to the NAT from the same source port it received the initial message from The initial SIP message sent to the proxy IP port initiates mapping on the NAT and the proxy returns packets to the NAT from that same IP port This is enabled in any NAT scenario Registering a client which is behind a NAT requires either a registrar that can save the IP port in its registration information based on the port and IP that it identifies as the source of the SIP message or a client that is 2000
155. server e Recursive follow me e g SIP account A forwards a call to SIP account B which in turn forwards it to SIP account C and so on e New in house developed B2BUA instead of Vovida for improved flexibility reliability and scalability e Modify CLI calling number for outgoing call e g to comply with vendor s ANI number format e Ability to browse a list of current calls on the PortaSIP server from the PortaBilling web interface e Flexible logic for RTP proxy dynamic proxy activation based on NAT status of the SIP UA e Support for video calls 2000 2006 PortaOne Inc All rights reserved www portaone com 6 Porta SIP PortaSIP User Guide Important upgrade notes We try to make the process of upgrading as easy as possible and to keep our releases backward compatible There are just a few things you should pay attention to when upgrading e Maintenance Release 11 introduces a new RTP proxy logic which is dynamically controlled from the billing side the RTTP parameter in porta sip conf is no longer being used To keep the same proxying mode for backward compatibility change the routing settings in porta billing conf as follows o RTTP 0 RTP proxy disabled RTPPLocalPolicy direct RTPPRemotePolicy direct o RITP 1 RTP proxy enabled for SIP SIP calls RTPPLocalPolicy all RTPPRemotePolicy direct o RTTP ALL RTP proxy enabled for all calls RTPPLO alPolicy all RTPPRemotePolicy all Note that the recom
156. server can try the alternative numbers until the call is answered GW NY 01 2 g QUE gt Porta KBilling Phone C 3 5 6 4 SIP phone A Phone X SIP phone R e C wishes to call A So he dials A s phone number since C is in the US he dials it using the North American format 2027810003 e The call is routed through the telecom network to gateway GW NY 01 When the incoming call arrives at the gateway 1 it is 2000 2006 PortaOne Inc All rights reserved www portaone com 28 Porta SIP System Concepts processed there in exactly the same way as a normal PSTN gt SIP call the number is transformed the call is authorized in the billing 2 and the timer starts to measure the maximum call time allowed based on A s current balance 3 The call is sent to PortaSIP 4 PortaSIP receives the INVITE but without authorization information So the PortaSIP server performs authorization in the billing based on the IP address and also requests billing assisted routing 5 PortaBilling recognizes that the destination is an account with follow me services enabled and produces a special list of routes o If the follow me mode chosen is When unavailable then a direct route to the account s SIP UA is included as the first route in the list with a default timeout o A list of follow me numbers is produced If the current time falls outside the specified period for a certain number it is removed from the
157. t 35132 created tag e97ded4c2b244b0cb4007cedb742 F088 21 Jul 23 37 26 1lefdc57b 80c3c2c 192 168 1 180 rtpproxy pre filling caller s address with 62 244 32 30 16384 21 Jul 23 37 26 GLOBAL rtpproxy sending reply 35132 70 68 0 213 INVITE is sent to the first gateway proxy in the route list 21 Jul 23 37 27 GLOBAL b2bua SENDING message to 195 234 212 1 5060 INVITE sip 380443333333 195 234 212 1 5060 SIP 2 0 Viat SIP 2 lt 0 UDP 70 68 0 213 5061 branch z9hG4bK6d42 63 featc5e4555efb2b88ac444e24 rport Max Forwards 70 From 758 lt sip 18667478647758195 234 212 1 gt tag qd5f2439660c74f6a92296794qd5f48q50 To lt sip 380443333333 195 234 212 1 gt Call ID 1l1efdc57b 80c3c2c 192 168 1 180 CSeq 200 INVITE Contact Anonymous lt sip 0 68 0 213 5061 gt Expires 300 User Agent Sippy Cisco GUID 2619723805 26646 8018 2210055510 2363514890 h323 conf id 2619723805 2664678018 2210055510 2363514890 2000 2006 PortaOne Inc All rights reserved www portaone com 79 Porta SIP Content Length 466 Content Type v 0 Setting up SIP Services application sdp o 2284 2284 IN IP4 192 168 1 180 s t 0 0 m audio 35132 RTP AVP 18 0 2 4 8 96 97 98 100 101 c IN IP4 70 68 0 213 a rtpmap a rtpmap a rtpmap a rtpmap a rtpmap a rtpmap a rtpmap a rtpmap a rtpmap a rtpmap a fmtp 101 0 15 a ptime 30 a sendrecv a direction active 0 PCMU 8000 4 0672378000 8 PCMA 8000 100 NSE 8000 1
158. tAuth Billing checks the destination number is it perhaps one of our SIP accounts Jul 21 22 37 26 CLD 380443333333 is not recognized as our account No this is an off net call Jul 21 22 37 26 Compute maximum call duration for account 18667478647758 with 9996029 96819 funds available Jul 21 22 37 26 Using peak rate since no off peak is defined Jul 21 22 37 26 PrepareNexecute GetPricePerDestination Jul 21 22 37 26 Maximum call duration unlimited announced as unlimited 5 0 1x30x0 1606 622417801x6x0 1606 0 by rate 125476 Maximum call duration is calculated according to available funds and rates Jul 21 22 37 26 Calculating routing for 380443333333 Jul 21 22 37 26 Looking up routes to 380443333333 using lt Default System Routing gt routing plan Jul 21 22 37 26 PrepareNexecute GetRoutingPerDestination Jul 21 22 37 26 Applying route outgoing CLD translation on 380443333333 Jul 21 22 37 26 Translation ele4 Co local gt lcco gt 1 ac gt 425 dp gt 1 ip gt 011 em gt 911 ex gt 411 cc 1 ac 425 dp 1 ip 011 em 911 ex 411 applied 380443333333 unchanged Jul 21 22 37 26 Result routes for 380443333333 Fastermination PSTN on node mera 195 234 212 1 priority 0005 cost 0 095 Free VoIP Network FVN VoIP to the remote GW 66 96 26 134 5061 priority 0005 cost 0 095 Apollo via SIP VoIP to the remote GW 67 105 130 10
159. taone com REFER 15 Porta SIP Setting up SIP Services a rtpmap 97 G726 24 8000 a rtpmap 98 G726 16 8000 a rtpmap 100 NSE 8000 a rtpmap 101 telephone event 8000 a fmtp 101 0 15 a ptime 30 a sendrecv a direction active a oldmediaip 192 168 1 180 21 JUL 23 37 26 GLOBAL b2Bua RECEIVED message from 0 68 0 213 5060 B2BUA receives this INVITE request INVITE sip 380443333333 70 68 0 213 5061 SIP 2 0 Record Route lt sip 70 68 0 213 ftag a4044d2fe886380800 1r gt Viet STE 2 0 UDE 70 68 0 213 branch z9hG4bK5e23 148da701768824ae9F2al 6b450895a5b 0 Via SIP 2 0 UDP 192 168 1 180 5060 rport 50563 received 62 244 32 30 branch z 9hG4bK 6ad8150b From 758 lt sip 18667478647758 70 68 0 213 gt tag a4044d2fe8 86380800 To lt sip 380443333333070 68 0 213 gt Call ID lefdc57b 80c3c2c 192 168 1 180 CSeq 101 INVITE Max Forwards 16 Contact 758 lt sip 18667478647758 62 244 32 30 50563 gt Expires 240 User Agent Sipura SPA2000 2 0 13 g Content Length 467 Allow ACK BYE CANCEL INFO INVITE NOTIFY OPTIONS REFER Supported x sipura Content Type application sdp PortaBilling Notify NAT v 0 o 2284 2284 IN IP4 192 168 1 180 c IN IP4 62 244 32 30 t 0 0 m audio 16384 RTP AVP 18 0 2 4 8 96 97 98 100 101 a rtpmap 18 G729a 8000 a rtpmap 0 PCMU 8000 a rtpmap 2 G726 32 8000 a rtpmap 4 G723 8000 a rtpmap 8 PCMA 8000 a rtpmap 96 G726 40 8000 a rtpmap 97 G726 24 8000 a rtpmap 98 G726 16 80
160. tems which attempt to preserve the RTP source port Linksys BEFSX41 Belkin F5D5230 4 natd bundled with FreeBSD 4 x and 5 x operating systems iptables bundled with Linux kernel 2 4 x i Commodity routers and NAT software bundled with popular operating systems which do not attempt to preserve the RTP source pott 1 Internet connection sharing software bundled with the Windows XP operating system 2 Netgear RP614 APPENDIX B Cisco GW Setup for PortaSIP COMEDIA Sip ua nat symmetric check media src APPENDIX C Clients Cisco ATA 186 Configuration for PortaSIP UIDO CLIENTS ACCOUNT ID PHONE NUMBER 1 PWDO CLIENTS PASSWORD FOR ACCOUNT ID 1 UID1 CLIENTS ACCOUNT ID PHONE NUMBER 2 PWD1 CLIENT S PASSWORD FOR ACCOUNT ID 2 GkOrProxy HP ADDRESS OF SERVER RUNNING PORTASIP Gateway 0 0 0 0 GateWay2 0 0 0 0 UseLoginID 0 LoginIDO 0 LoginID1 0 AltGK 0 0 0 0 AltGKTimeOut 0 GkTimeToLive 300 2000 2006 PortaOne Inc All rights reserved www portaone com 1 93 Porta SIP Appendices GkId UseSIP 1 SIPRegInterval 8 MaxRedirect 5 SIPRegOn 1 NATIP 0 0 0 0 SIPPort 5060 MediaPort DIFFERENT FOR EACH CLIENT AS DESCRIBED IN THE SETUP GUIDELINES OutBoundProxy 0 0 NatServer HP ADDRESS OF SERVER RUNNING PORTASIP NatTimer Oxle LBRCodec 0 AudioMode 0x00150015 RxCodec 0 TxCodec 0 NumTxFrames 1 CallFeatures Oxf PaidFeatures Oxf EEEE CallerIdMethod 0xc0019e60 Feature
161. the User Agent and Contact fields contain some values These fields will show the account s current registration information E 4 Account Info Retail Customer EasyCall Ltd gt save i Save amp Close cos gi Rate Lookup to gout Bio Account ID 12061234570 Product USD SIP Subscribers Y Blocked Fi Balance 0 00000 USD User Agent Sipura SPA2000 2 0 13 g Contact 5ip 12061234570 70 68 0 213 5060 Account Info Maintenance Subscriber Additional Info Life Cycle User Interface Notepad Customer EasyCall Ltd Credit Limit usD Type Credit Opening Balance 0 00000 USD VoIP Password lavengio Refunds 0 USD E mail Non call related charges 0 USD Batch easycall Control number 4 4 Try to make a call using one of the accounts 5 Browse the SIP server log file var log sipenv lt sipserverIP gt log sip log on the SIP server host Some of the SIP auxiliary requests or parameters have been removed for greater clarity You can also browse the SIP log file from the PortaBilling web interface Choose the SIP Log Viewer item from the main menu and type in the call id for this call If you do not know the call id you can display a list of call attempts for the last 5 minutes or so and find the call id for your call SIP user agent attempts to make a call via the SIP server 21 Jul 23 37 26 GLOBAL ser RECEIVED message from 62 244 32 30 50563 INV
162. their respective owners 2000 2006 PortaOne Inc All rights reserved www portaone com 1 Porta SIP PortaSIP User Guide Table of Contents PAR cate eee ee pa ats s ena deo ce cesarean AAE eee 4 Hardware and Software Requirements 0 cece 5 I FUSS TOV AEE seu cette cheeses eh ct os octets ste E EEEE 6 What s New in Maintenance Release 117 6 Important upgrade notes in 7 1 System Concepts ss 8 PortaSIP s Role in Your VolP Network 9 PortaSIP CCMPONME NES ennemies 11 Call Process Supported Services 12 VAS LPS E E E 21 Clustering Of PortaSiP servers ins 22 Clustering of PortaSl P Servers 22 Call flow scenarios for a PortaSIP cluster 24 Advanced Features enr 28 Understanding SIP Call ROUtINg hu 33 NAT Traversal Guidelines oo ccccsesescsescscssssscsssscsessscsssssssesssssessssssesesessseeen 41 Auto provisioning of IP Phones ccc cscsescscsessssssssssssstesesssessseneees 47 2 Setting up SIP Services 50 BASIC SIP SeNi C ee 51 Setting up auto provisioning of IP Phones cece 94 3 HOWTO ER UT 103 configure my Cisco gateway to accept incoming SIP calls and terminate them to a telephony network 104 configure my Cisco gateway to send outgoing calls using SIP 105 configure my Cisco gateway for PSTN gt SIP Service oo eee 106 SUpport incoming H323 and SIP calls on the same gateway 106 configure m
163. ting up SIP Services 3 You can browse current configuration parameters on the expert screen Mee G e Cisco ATA Expert gt Gd save E save cie 195 140 247242 9933 M Logout Version 3 1 1 atasip Build 0406294 IP 195 140 247 242 3333 MAC 0 9 232 142 49 238 00 09 E8 8E 31 EE Mode v w ve accounts codecs ec orma ran Debus sP cal reatures FAS Timeouts Ringe Tones Provisoning Othe DHCP Do Not Request Option 150 Request Hostname IP Netmask Default Route d O Use DHCP Server Supplied DNS Primary DNS Secondary DNS 0 0 0 0 0 0 0 0 IP precedence ToS bit of UDP packets Request High Reliability Request High Throughput Request Low Delay Datagram Precedence d O Fi ow VLAN Use VLAN IP Encapsulation g TCP CoS 802 1P priority UDP CoS 802 1P priority User specified 802 1 VLAN ID 802 1Q ID o Mes WY 4 Version 3 1 1 atasip Build 0406294 IP 195 140 247 242 3333 LBR Codec RTP Frame Size ms TxCodec Preference RxCodec Preference Cisco ATA Expert gt Te Save E saves cose 195 140 247 242 9933 M Logout SIP M MAC 0 9 232 142 49 238 00 09 E8 8E 31 EE Mode Send A Ringback Tone To The Caller Mix Audio And Call Waiting Tone During A Call Phone 1 G 711 Silence Supression G 711 Only DTMF Relay Hookflash Relay o zo M oO by negotiation VW G
164. tion to handle PSTN gt SIP calls You can create this application yourself according to the functionality description in this guide PSTN2SIP application may be purchased from http store portaone com configure SIP phone X made by vendor Y Obviously we cannot provide a sample configuration for every possible SIP phone model Please check the documentation shipped with your device Essentially however you need to configure the following settings e IP address of the SIP proxy IP address or hostname of the PortaSIP server e CID Caller Identification e Login and password account ID and password of the corresponding account in PortaBilling 2000 2006 PortaOne Inc All rights reserved www portaone com 108 Porta SIP How to e Preferred audio codec depends on your network characteristics should be compatible with the codec used by other components e g VoIP gateways used for PSTN termination In the case of PortaSIP both the login name and CID should be set to the same value Set the preferred audio codec to G 723 if your phone supports this Likewise enable in band alerting if your phone supports it as this will help in situations when the phone is behind a NAT bill SIP to SIP calls By default calls from one SIP account to another are treated as on net ones and are therefore not billed However if you want to bill your customers for such calls you can do the following
165. tomatic disconnect of calls when one of the parties goes offline due to a network outage e Various IP Centrex features call waiting call hold music on hold abbreviated dialing follow me etc e Fail over routing a list of routes arranged according to cost preference and customer routing plan is supplied by PortaBilling100 e Forwards calls to the unified messaging service PortaUM if a SIP phone is not available 2000 2006 PortaOne Inc All rights reserved www portaone com 1 0 Porta SIP System Concepts PortaSIP Components SIP UA SIP UA D Porta 4 Billing100 ee ans in oa RADIUS Client auus SIP Express Router SER Li su o a RADIUS z Client s CRE er Agent Server RADIUS rate RTP SIP Porta Proprietary PortaSIP components e SIP Proxy Server SIP Express Router on the diagram The SIP Proxy Server performs a number of functions such registering SIP telephones dealing with NAT issues etc e Back To Back User Agent B2BUA The B2BUA SIP based logical entity can recetve and process INVITE messages as a SIP User Agent Server UAS It also acts as a SIP User Agent Client UAC determining how the request should be answered and how to initiate outbound calls Unlike a SIP proxy server the B2BUA maintains the complete call state Integrating B2BUA with PortaSIP ensures that every call made between endpoints
166. tor only v General Info Breakage lo Maintenance Period none Maintenance Fee Maintenance Charging Account default ACL Account self care M Default Discount Plan None M Info URL Description Product for SIP users no monthly fees E 4 Edit SIP Subscribers Product Product Name 51P Subscribers 7 Currency USD o Managed By Administrator only ad General Info Online Signup Info Online Signup Accessibility Notepad af Edit Hode Access Code Info Digits Tariff Delete DenoSIP Node x ANY usD SIP Phone subscribers E r E p E 4 Edit SIP Subscribers Product gt Add i save fal save amp close Close 15 Ratetookup Dieu Bio Product Name SIP Subscribers Currency USD Managed By Administrator only General Info Online Signup Info Online Signup Accessibility Notepad Edit Hode Access Code Info Digits Tariff Delete ANY ANY USD GlobalNet DemosiP Node SIP Phone subscribers In the Management section of the Admin Index page choose Products 1 On the Product Management page click the Add icon 2 Fill in the Add product form o Product name product object name o Currency product currency only tariffs which have the same currency will be permitted in the accessibility list 2000 2006 PortaOne Inc All rights reserved www portaone co
167. tric check media src have problems when trying to use SIP phone X made by vendor Y with PortaSIP Unfortunately not all of the many SIP phones available on the market today fully comply with the SIP standard especially low end products We use Cisco ATA 186 as a reference phone and the Cisco ATA PortaSIP combination has been thoroughly tested If you are unable to get your third party vendor SIP phone working properly follow the instructions below e Make sure the phone has been configured properly with such parameters as account ID password SIP server address etc Consult the product documentation regarding other configuration settings e Check the PortaSIP and PortaBilling logs to ensure that there is not a problem with the account you are trying to use for example an expired or blocked account e Connect the Cisco ATA or Sipura to the same network as your SIP phone If possible disconnect the SIP phone and use the same IP address for the Cisco ATA Sipura as was previously used by the third party SIP phone Configure the Cisco ATA Sipura with the same account as was used on your third party SIP phone e Try to make test calls from the Cisco ATA Sipura e If you have followed the preceding steps and the problem disappears then this means your third party vendor SIP phone ts not working according to the standard Contact the vendor of the SIP phone and describe the problem e If this problem with the Cisco ATA S
168. ts and routing from a single location on the web interface Thus when such a routing tariff is associated with a connection you can send calls for termination to all prefixes for which rates exist in the tariff Multiple routes It is dangerous to have only one termination partner if it is down your customers will not be able make any more calls Normally you will try to find several vendors and enter their rates into the system Each connection to a vendor with routing tariff will produce one possible route and PortaBilling will arrange them according to cost or your other preferences Routing plans Routing preferences in the rate allow you to specify that for example you would rather send a call to MCI than to T Systems However this decision is global and so will apply to all calls made in your system But what if you would like to use MCI first for customer A while T Systems should be the first route for customer B and customer C should be routed to MCI only This can be accomplished using routing plans A routing plan defines the routes for which categories are available as well as in which order they should be arranged For instance in the example above MCI may be assigned as the Normal route category and T Systems as the Premium category After that three routing plans will be created e Quality includes first Premium and then Normal routing categories e Ordinary includes first Normal and th
169. ult web host so any IP phone can access its configuration file just by sending a request to the http port of the PortaBilling100 slave server For phones that do not support http provisioning you may use the TFTP protocol The main disadvantage of TFTP is that it has a higher chance of being blocked by a firewall There is no support for subdirs in the TFTP provisioning server path so all provision configs are stored without a env subfolder in howe porta admin apache htdocs unlike other UA profile configs You may change the default location for storing configuration files 1 To change the http root dir edit the DocumentRoot parameter for port 80 in the Apache host s configuration file usr local etc apache porta httpd conf lt VirtualHost default 1602 DocumentRoot home porta admin apache htdocs Options ExecCGI DirectoryIndex index pl lt VirtualHost gt Note If you change this value you must make corresponding changes in an additional list of configuration files Do not forget to restart the Apache server afterwards 2 To change the PortaBilling output directory for compiled profiles edit the section UA_Profiles ResultDir parameter in the usr home porta admin etc porta admin conf file 3 To change the PortaBilling result dir for intermediary non yet compiled text files edit the section UA_Profiles Dir parameter in the usr home porta admin etc porta admin conf file 4 To change the tftp server ro
170. umber e A list of possible termination addresses will be produced this will include remote IP addresses for the VoIP connections and IP addresses of your own nodes with telephony connections e This list will be sorted according to the routing preference with entries after the first huntstop being ignored e lt A list of these IP addresses with optional login and password for SIP authentication will be returned to the SIP server NAT Traversal Guidelines NAT Overview The purpose of NAT Network Address Translation is to allow multiple hosts on a private LAN not directly reachable from a WAN to send information to and receive it from hosts on the WAN This is done with the help of the NAT server which is connected to the WAN by one 2000 2006 PortaOne Inc All rights reserved www portaone com 41 Porta SIP System Concepts interface with a public IP address and to the LAN by another interface with a private address This document describes issues connected with the implementation of NAT and its implications for the operation of PortaSIP with an overview of some fundamental NAT concepts The NAT server acts as a router for hosts on the LAN When an IP packet addressed to a host on the WAN comes from a host on the LAN the NAT server replaces the private IP address in the packet with the public IP address of its WAN interface and sends the packet on to its destination The NAT server also performs in memory mapping b
171. under a NAT from to a VoIP GW and the remote gateway supports SIP COMEDIA extensions In this case the RTP stream may flow directly between the gateway and the SIP phone and there is no need to use an RTP proxy However you need to configure your Cisco GW as per APPENDIX B Cisco GW Setup for PortaSIP COMEDIA in order to ensure proper NAT traversal 3 A call is made from to a UA under a NAT from to a VoIP GW and the remote gateway does not support SIP COMEDIA extensions An RTP proxy is required in this case In appendices A through C you will find a list of tested routers as well as a typical configuration for Cisco IOS software and Cisco ATA 186 telephones which has been adapted for optimal NAT traversal performance 2000 2006 PortaOne Inc All rights reserved www portaone com AG Porta SIP System Concepts PortaOne RTP proxy This provides an effective NAT traversal solution according to the RTP proxy method described above The RTP proxy is fully controlled by PortaSIP and is absolutely transparent to the SIP phone The RTP proxy does not perform any transcoding and so requires a minimum amount of system resources for call processing A single RTP proxy on an average PC server can support about 5 000 simultaneous calls PortaSIP can detect whether an IP phone is behind a NAT thus employing the smart method of RTP proxying only activating it when necessary Of course you have control over the RTP proxy
172. visioned by PortaSwitch First of all you must make sure that your device supports auto provisioning see APPENDIX G SIP devices with auto provisioning Then create the required IP phone profile and enter information about the IP phone into the inventory Provision the SIP service as described in this manual and then assign it to an available port on your IP phone in the account info screen for a SIP account Enter information about the provisioning server into your IP phone s configuration In some cases you may need to restart the IP phone in order to force a configuration update from the provisioning server 2000 2006 PortaOne Inc All rights reserved www portaone com 1 1 1 Porta SIP Administration FAQ 4 Administration FAQ 2000 2006 PortaOne Inc All rights reserved www portaone com 112 Porta SIP Administration FAQ Troubleshooting Common Problems No or one way audio during SIP Phone SIP Phone calls This problem usually means that one or both phones are behind a NAT firewall Unfortunately unless the RTP Proxy is turned on or certain smart SIP phones NAT routers are used there is no way to guarantee proper performance in such cases see Nat Traversal section for details One way audio during SIP Phone Cisco gateway calls This problem can occur if the Cisco GW is not configured properly Please check that the GW contains the following in its IOS configuration Sip ua nat symme
173. y Cisco ATA186 to work with PortaSIP 107 provide services to and bill a customer who has a SI P enabled gateway but no authorization capability e g Cisco AS5350 107 Make all SIP calls to a certain prefix NNN go to my gateway XXX 107 allow my customer to have two phone numbers from different countries which will both ring on the same SIP phone 108 Create an application to handle PSTN gt SIP Calls woe 108 configure SIP phone X made by vendor Y NL 108 e DISPOSER ee eee 109 bill incoming calls from PSTN to SIP using a special rate 109 provide error messages from the media server in my users local AO OS omen en EEE EN SPH Nr Sta oR aro OR na nT ee em 110 calculate how much bandwidth need for my PortaSIP server 110 enable my SIP phone or ATA to be automatically provisioned by POrtA MCU acai Seat tected Siena nod EAE EE ae erence 111 4 Administration FAQ 112 Troubleshooting Common Problems esse cs csesesescseses essen 113 a O 114 2000 2006 PortaOne Inc All rights reserved www portaone com 9 Porta SIP PortaSIP User Guide PortaSIP configuration iii 118 Appendices a ere re 122 APPENDIX A Tested Routers and NAT Software 123 APPENDIX B Cisco GW Setup for PortaSIP COMEDIA 123 APPENDIX C Clients Cisco ATA 186 Configuration for PortaSIP 123 APPENDIX D Client s Sipura Configuration for PortaSIP 125 APPENDIX E Confi
174. you may define a SRV record describing the available SIP servers sip udp proxy SRV 10 0 5060 portasipl SRV 10 0 5060 portasip2 SRV 60 0 5060 portasip3 2000 2006 PortaOne Inc All rights reserved www portaone com 27 Porta SIP System Concepts The first two servers have a higher priority 10 so they will be tried first Also note that DNS SVR allows you to specify which port should be used for communication On your SIP phone you should specify the following SIP proxy registrar proxy mysipcall com Use DNS SRV yes DNS SRV Auto Prefix yes If you do not switch on the auto prefix feature then the SIP proxy address must be entered as_sip udp proxy mysipcall com So now when a SIP phone is switched on it will first query the DNS database for servers for _sip_udp_ proxy mysipcall com receiving a list of recommended servers portasipl mysipcall com portasip2 mysipcall com and portasip3 mysipcall com After that it will obtain the IP addresses of these servers from the DNS database and attempt to contact them in sequence until it succeeds Advanced Features Follow me services Due to the volatile nature of VoIP networks the customer may wish to use standard PSTN calls as a backup He can define a list of follow me numbers for each of which a period of validity can be defined e g he wants to receive calls to his mobile phone only from 8am to 9pm When a call arrives on his original SIP account the SIP

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